| cfa5bf04ff250a14b11a75bcfa35ee2d6dbab4e9 |
07-Jan-2015 |
Chong Zhang <chz@google.com> |
Merge "make libserviceutility a shared lib" into lmp-mr1-dev
|
| 5db7897ddd32e3ec3ab45ecdb6f21b6265e7e14a |
07-Jan-2015 |
Chong Zhang <chz@google.com> |
make libserviceutility a shared lib
so that we have only one getpid_cached in mediaserver process
bug: 18919657
Change-Id: Iff3cd932c9110e874b3885f79705f49bf3e3f1fc
ndroid.mk
|
| 610255e8f2d02cb3513695eacf1c37d66b3553d6 |
07-Jan-2015 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: pause HW A/V sync output when AudioTrack underruns" into lmp-mr1-dev
|
| e443ffac2a09b69470c4663ee031a527c53738d9 |
05-Jan-2015 |
soon1.choi <soon1.choi@lge.com> |
DO NOT MERGE - fix memory leak in PatchPanel
Before the audio patch is created, audio patch with same handle is
removed from vector for patches. At this time, Patch allocated in the
heap is not freed. This causes memory leaks in the mediaserver.
Bug: 18904322.
Change-Id: Id0d28c36a7e7fbf473753bf4ead7518f28c1b998
atchPanel.cpp
|
| 0f7b5f2b231caf87da9b20b74d086e5a9d6f4a9d |
19-Dec-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: pause HW A/V sync output when AudioTrack underruns
Do not standby, starve or feed 0s to the audio HAL on direct
output using HW A/V sync mode.
Bug: 17883772.
Change-Id: I11e6c97ec24360d75f9b602814d40a54b60cb7a7
hreads.cpp
hreads.h
|
| 145cf5d885c238b690ab8240137934327b85c427 |
17-Dec-2014 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: implement pause/resume for direct outputs" into lmp-mr1-dev
|
| 97bb33f58d742539f3382583d7978fca71ffa2d5 |
13-Dec-2014 |
Jean-Michel Trivi <jmtrivi@google.com> |
Fix permission check for audio recording
Define input types covering the different usecases for audio
recording, and check the corresponding permissions when
starting to record.
Move the permission check from audio flinger to audio policy,
as only the policy has the information to determine which
permission to enforce.
Fix missing permission when an application records audio
and the audio is injected by an external policy.
Bug 18736417
Change-Id: If7ec040502242c990ac8ea464db484339bdce573
udioFlinger.cpp
|
| d1f69b0b17acbd96987ecb2f3378abd394d05903 |
15-Dec-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: implement pause/resume for direct outputs
Extend pause/resume support to direct output threads
(was only for offload threads).
If the HAL implements pause/resume, track pause/resume is forwarded to
the HAL.
Pause, flush, resume sequence is respected by executing the HAL
calls in the playback thread (same as offload).
Make sure the track flags on client side are consistent with the
flags on server side.
Bug: 17883772.
Change-Id: I89b360d69818f7a9204bd36e3ec63a79e106ecf1
hreads.cpp
hreads.h
racks.cpp
|
| 94a92c69af528edf6ec17d7978a0c3bb6ab51e63 |
11-Dec-2014 |
Phil Burk <philburk@google.com> |
Merge "[audio][audioflinger] Consume all frames when a DirectThread is done." into lmp-mr1-dev
|
| 99adee3c3d9cde6819741a38163954808fea270a |
11-Dec-2014 |
Phil Burk <philburk@google.com> |
[audio][audioflinger] Consume all frames when a DirectThread is done.
This bug was causing movies to think there was still data left to play so they never finished.
Bug: 18360054
Change-Id: If7aee81d62b7af8532abd14579677848a06fe2b1
Signed-off-by: Phil Burk <philburk@google.com>
hreads.cpp
|
| 275e8e9de2e11b4b344f5a201f1f0e51fda02d9c |
01-Dec-2014 |
Eric Laurent <elaurent@google.com> |
audio policy: add support for custom mixes
Add support for custom mixes in AudioPolicyManager.
Two methods are added to register or unregister a list of custom mixes
with their attributes and format.
getOutputForAttr() and getInputForAttr() first look for a match in
registered mixes before defaulting to normal output/input selection
Remote submix device connection disconnection now takes address into
account to identify the correspnoding custom mix.
Bug: 16009464.
Change-Id: I3f1c2a485a0fb71b1f984ed0adc9b68aa971e408
hreads.cpp
|
| 9a1c8892f6835325db6931529dc74bf98cf0aee8 |
03-Dec-2014 |
Andy Hung <hunga@google.com> |
Fix getTimestamp if offload thread is not initialized.
Returns INVALID_OPERATION.
Bug: 18605649
Change-Id: I440fb687fbf9249098e049982fa83bcd1515fd80
hreads.cpp
|
| e83b55dc29ca16092ba02f36f55fa6e0e37fd78c |
14-Nov-2014 |
Eric Laurent <elaurent@google.com> |
audio policy: new getOutputForAttr() prototype.
Update getOutputForAttr() prototype and group all logic
dealing with audio attributes to stream type conversion in
audio policy manager.
getOutputForAttr():
- specifies the audio session (for future use)
- returns a status code
- receives either stream type (for legacy) or audio attributes
- returns an updated streamtype
Remove logic dealing with legacy stream types to attributes conversion
from AudioTrack.
Use correct type for audio sessions in other APIs (startOutput() ...).
releaseOutput() specifies the audio session (for future use).
Bug: 18067208.
Change-Id: I1bfbe9626c04c7955d77f8a70aecfad2cb204817
hreads.cpp
racks.cpp
|
| 223fd5c9738e9665e495904d37d4632414b68c1e |
11-Nov-2014 |
Eric Laurent <elaurent@google.com> |
audio: new routing strategies and stream types
Added new routing strategies and stream type for internal use
by audio policy manager and audio flinger:
- One for accessibility to allow different routing than media
- One for re-routing (remote submix) in preparation of dynamic
policies
- Added stream type for "internal" audio flinger tracks used
for audio patches and duplication.
Bug: 18067208.
Change-Id: I88f884b552e51e4a49c29125e5a1204cf58ff434
udioFlinger.cpp
udioFlinger.h
hreads.cpp
hreads.h
racks.cpp
|
| ec0eeafa8a7fd882e8206e9cf8c4c7e1db81857b |
20-Nov-2014 |
Haynes Mathew George <hgeorge@codeaurora.org> |
Fix format conversion in DuplicatingThread
DuplicatingThread converts PCM data from the format of the source thread
to PCM_16BIT before writing to all connected output tracks.
This needs to be done just once before writing to all tracks.
Bug: 18461844
Change-Id: If05254f87f9198e09a0fa3c4e857afbb53ae7073
hreads.cpp
|
| 27876c02c21cd3ab7ef01bdd1fc5400c5143584a |
10-Sep-2014 |
Andy Hung <hunga@google.com> |
Prevent stopped audio tracks from mixing data
Static tracks should have zero framesReady when stopping or stopped,
to prevent mixer from running potentially indefinitely with loop mode.
Bug: 16070883
Change-Id: Iab9968044d1a6f8543bbcbfb246f49f27d68cace
racks.cpp
|
| 02b5708776ba2a9b4ff8c09008483aef7dbe38c7 |
08-Nov-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix duplicating thread output buffers clear
Also clear mix buffer in DuplicatingThread::threadLoop_mix()
instead of just sink buffer when output threads are not
ready.
Bug: 18262127.
Change-Id: I4c22bbeaccdcc8d7c10c5882dde3439eb73d6ede
hreads.cpp
|
| b10352f4bc11ae6f3d93e464074709555ecf9e06 |
04-Nov-2014 |
Eric Laurent <elaurent@google.com> |
remove some restrictions on effect output channels
Do not prevent from attaching effects to a non stereo
output thread, unless the output thread is a mixer thread.
Bug: 18157592.
Change-Id: I6ac3187187a1b8aade7db04ea6dfbc47dacc25c3
udioFlinger.cpp
|
| fd4c14883b268a0bc5514da135fe6b7d1ce2071b |
07-Aug-2014 |
Jean-Michel Trivi <jmtrivi@google.com> |
Always use an address for remote submix
Usage of remote submix device for audio rerouting (e.g. wifi display)
didn't mandate the use of addresses. Use "0" as the default address
when none is specificed.
In logs, only use hex format for audio devices
Bug 16009464
Change-Id: Ibfb1ce6881eba8b7e34420293b8a7077a6e659e6
udioFlinger.cpp
|
| 9d0d5958969cffae5d4305025bc20d05f32e6ff2 |
22-Oct-2014 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: new HW A/V sync ID allocation flow" into lmp-mr1-dev
|
| fa90e84ed0cb2abdc0e0873a06d10ccf2b3c34f6 |
18-Oct-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: new HW A/V sync ID allocation flow
The HW A/V sync ID is now allocated by the audio HAL before the
output stream is created by a call to global get_parameters() with
key AUDIO_PARAMETER_HW_AV_SYNC.
When the AudioTrack is created, the HW A/V sync ID is communicated
to the output stream by stream set_parameters() with key
AUDIO_PARAMETER_STREAM_HW_AV_SYNC.
Bug: 17112525.
Change-Id: Ia8bc6f3bf9f358aa89f3f56ac554e893a19811ad
udioFlinger.cpp
udioFlinger.h
|
| 36d9256f956b2ed4d7a221d84ad05bf4ba482958 |
20-Oct-2014 |
Marco Nelissen <marcone@google.com> |
am ac302143: Fix the fix
* commit 'ac302143551a8b964f026385a524dda9ff8ea5ba':
Fix the fix
|
| ac302143551a8b964f026385a524dda9ff8ea5ba |
20-Oct-2014 |
Marco Nelissen <marcone@google.com> |
Fix the fix
We need to clear the buffer a little more often.
Bug: 18023017
Change-Id: If8a522b9133a4269e2435af8958f6ced322e71e9
hreads.cpp
|
| 145a4c679a9ab27aceaa1d386ea9aca1c0cd4d33 |
20-Oct-2014 |
Marco Nelissen <marcone@google.com> |
am 22bc10a3: Merge "Clear effect buffer before sending to chain" into lmp-dev
* commit '22bc10a347569a0ab2980c24783e31aba4d7414d':
Clear effect buffer before sending to chain
|
| 57088b5c8e76855b99b3e6b3e410de5b6382670e |
18-Oct-2014 |
Marco Nelissen <marcone@google.com> |
Clear effect buffer before sending to chain
otherwise the effect might get a buffer with previous data instead
of silence.
Bug: 18023017
Change-Id: Ic6be5ce0e74552aac2a0cce9c660893f9ad27f5c
hreads.cpp
|
| f53eaf49df1e1e1c01708919e9e0a046543de9b6 |
09-Oct-2014 |
Hochi Huang <hochi.huang@mediatek.com> |
fix FM no sound issue
Review: https://partner-android-review.git.corp.google.com/#/c/184276
Signed-off-by: Carson Liao <carson.liao@mediatek.com>
Change-Id: I0847c8a6e6d054a9258b52402c914105d902c573
erviceUtilities.cpp
erviceUtilities.h
|
| 2b217bb3aee87ce8486014f261c0f498f6209e80 |
03-Oct-2014 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: fix pre processing transfer between record threads." into lmp-dev
|
| 1b92868010b5c1409692a86f6b27e4a265b64c1a |
03-Oct-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix pre processing transfer between record threads.
Fix two problems remaining with pre processing effects transfer from
one record thread to the next in case of tear down due to device connection:
1 - the enabled state of the effects was not communicated to the new HAL
input stream.
2 - the effects saved in orphan chains list were not transfered to the
new thread when a AudioRecord was created.
Bug: 17757378.
Change-Id: I0923c98470db3b51154dc89846157780a4c21e86
udioFlinger.cpp
ffects.cpp
ffects.h
hreads.cpp
|
| 4c053ea158b29fa2cdd4c6f39d3c8da4ee5a7a02 |
28-Sep-2014 |
Glenn Kasten <gkasten@google.com> |
Fix uncertainty of one normal mix buffer in AudioTrack::getTimestamp
The per-thread timestamp latch was not synchronized with the per-track
released frames. Now the value of each track's released frames is
latched along with the timestamp.
Bug: 17531839
Bug: 17669342
Change-Id: I9d50c8c6a5de55a3f4561ac40e20d497376c1257
hreads.cpp
hreads.h
racks.cpp
|
| cb4b6e9f05d1c0b3fcc5be5ebe3f7d6f32669b1a |
01-Oct-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix failure to indicate volume to effect
If an effect is added and enabled after a track is started
it will not receive current volume information because volume
is indicated only to active effects and if there is a volume change.
Add a flag to force EffectChain::setVolume_l() to pass volume indication
to all effects each time an effect is enabled.
Bug: 17572556.
Change-Id: I6b68488c9aca8b2095fb210aa124d02e4f759660
ffects.cpp
ffects.h
|
| a49c94ea49c8329a1bdefe980e1f7d335c0d1ec5 |
02-Oct-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Revert "Fix uncertainty of one normal mix buffer in AudioTrack::getTimestamp"" into lmp-dev
|
| 6954127b7ace022677ac407ff943c2793f8a11be |
02-Oct-2014 |
Glenn Kasten <gkasten@google.com> |
Revert "Fix uncertainty of one normal mix buffer in AudioTrack::getTimestamp"
This reverts commit 54464ba861aaafd11ee5645f5d1ecd1171c6e9fe.
Bug: 17733142.
Change-Id: I9886d40032b2f7d96bee03e600e14f78943d8052
hreads.cpp
hreads.h
racks.cpp
|
| 955e24d3a8e218d4711cabc6558781e095011132 |
30-Sep-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix uncertainty of one normal mix buffer in AudioTrack::getTimestamp" into lmp-dev
|
| f71224ce266abb39f83ecba87783c2347631654a |
30-Sep-2014 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: forward flush to direct output streams" into lmp-dev
|
| 54464ba861aaafd11ee5645f5d1ecd1171c6e9fe |
28-Sep-2014 |
Glenn Kasten <gkasten@google.com> |
Fix uncertainty of one normal mix buffer in AudioTrack::getTimestamp
The per-thread timestamp latch was not synchronized with the per-track
released frames. Now the value of each track's released frames is
latched along with the timestamp.
Bug: 17531839
Bug: 17669342
Change-Id: I96252782911bd42a2534cc4030521b7eeb99a7b9
hreads.cpp
hreads.h
racks.cpp
|
| e659ef420dae0caae84ab78f9df8952acb9ad3a0 |
29-Sep-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: forward flush to direct output streams
Track flush should be forwarded to the audio HAL when playing
on a direct output stream.
Bug: 17704953.
Change-Id: Iafc94327fe9aebf1d4eaa97b76c6767b7ac5babe
hreads.cpp
hreads.h
racks.cpp
|
| ba0b34c18da93681c0813ecdab19b0e215b6d261 |
28-Sep-2014 |
Glenn Kasten <gkasten@google.com> |
Fix tee sink
Note: this does not fix the bug below, but it helps to investigate the bug.
Bug: 17578238
Change-Id: I0fd27a477b73ba0e1c5a658bdfad17c500f5dc67
hreads.cpp
|
| 80b617cf82e0030deef4f968a6341e2867166135 |
17-Sep-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: remove debug log.
Remove debug log left by commit ab5cdbaf.
bug: 17537746.
Change-Id: I10f672f607f8e87caaa8228fc05eff8aee7dfa15
hreads.cpp
|
| cf7863ea8d9137aadf6bfd9756eb07ebd1c81b5c |
15-Sep-2014 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: fix pre processing effect leak" into lmp-dev
|
| aaa44478a373232d8416657035a9020f9c7aa7c3 |
13-Sep-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix pre processing effect leak
When a capture thread was closed, the effects attached to this thread
were left dangling and the associated effect chain destroyed.
When their last client was disconnected, the effects were not released
properly from the effect library because the destruction process could
not be completed without the effect being attached to a thread.
A similar problem prevented a RecordTrack to be properly released if
its client was destroyed after the capture thread.
The fix consists in allowing the effect or record track to be properly
released even if its parent thread cannot be promoted.
Also save any effect chain still present on a closed capture thread
in case a new client wants to reuse the effects on the same session later.
Bug: 17110064.
Change-Id: I5cd644daa357afd1f3548f9bcb28e6152d95fdb8
udioFlinger.cpp
udioFlinger.h
ffects.cpp
ffects.h
hreads.cpp
hreads.h
rackBase.h
racks.cpp
|
| 2f035f59d1e28728d38d18a7f0f7a9c6e8b0c11b |
14-Sep-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: send mic mute command to all audio HALs
Mic mute should be sent to all audio HALs, not
only the primary HAL as telephony can use
capture devices on other HALs (e.g USB)
Bug: 17321604.
Change-Id: I658f6084d5b5cdc5a70784661d5cea0b6f81c3a9
udioFlinger.cpp
|
| f8fd8d6daab5ced86ce950be68070d27c24d487f |
10-Sep-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix audio patch release
Fix error in PatchPanel:releaseAudioPatch() causing
the wrong patch handle to be passed to the audio HAL.
Bug: 17368621.
Change-Id: I037fd2e6273508c830414b5900d07c0178b96dae
atchPanel.cpp
|
| 3abc2ded40066f3b1df23aceb553f22d569c5cd3 |
06-Sep-2014 |
Glenn Kasten <gkasten@google.com> |
Make IPowerManager native conform to .aidl for oneway
But provide a temporary escape hatch for AudioFlinger.
This oneway option will be removed as soon as possible.
Bug: 16408906
Change-Id: I058a50906af810787e444a96819cb781b7f639ad
hreads.cpp
|
| 23c8cf958b1a756d5e0c78c46eac676676be4fdd |
03-Sep-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Restore the ability to build tee sink for debugging" into lmp-dev
|
| 97d547da43c9c41711d1ed1e3f4fa87c2ee3cb9a |
02-Sep-2014 |
Eric Laurent <elaurent@google.com> |
audio flinger: fix use of global effect buffer
When float mixing format is enabled, audio effects use a specific
buffer in 16 bit format. This buffer was not enabled when only
global effects were present resulting in the mixer accumulating directly
to the output sink buffer.
Bug: 16202259.
Change-Id: I3537d36c63196d97772f0ab22dd31d04dfa7fdd6
hreads.cpp
|
| 329f6511ee4e03a4605c70bbda8d3a96d2544884 |
29-Aug-2014 |
Glenn Kasten <gkasten@google.com> |
Restore the ability to build tee sink for debugging
This changelist does not enable tee sink, but makes it possible to do so.
Tee sink had suffered some bit rot since it is not built by default.
Also fixes a crash for > 2 byte per sample or > 2 channels.
Still does not write correct header for floating-point;
that will be best solved by moving to libsndfile.
Bug: 16990102
Change-Id: I8e92c588ccc513d7802d696fcfb324e815772df6
udioFlinger.cpp
astMixer.cpp
racks.cpp
|
| 395db4bfa5b43a839f95632676d59cde99a9840d |
26-Aug-2014 |
Andy Hung <hunga@google.com> |
Fix AudioMixer MONO muted track handling
Incorrect handling caused "stereo" mode to be used.
Augmented error handling to print more debug info.
Bug: 17208475
Change-Id: I5bcaf0ba5bd5a4f7d4885f8cce43bbf42e5f1e6b
udioMixer.cpp
|
| 735f45fbc37d7905ffb722f40727edbed82319b7 |
19-Aug-2014 |
Glenn Kasten <gkasten@google.com> |
Fix memory access error due to race during fast track removal
Bug: 16205169
Change-Id: Ie15ec2a737b6ddd16b2d26030d01564ff70177f5
hreads.cpp
|
| 49d00ad9164ea5ce48c85765a2b6460d9b457d38 |
21-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
Fix overruns when capturing with resampling
Bug: 16658148
Change-Id: I2dd04b23a143e0948b299d408790e7a46b5a53b0
hreads.cpp
|
| e7a1737c92cf2e84754ffbc52cfec8edeffcbc75 |
12-Aug-2014 |
Ricardo Garcia <rago@google.com> |
Merge "Fix to support mono in effect buffer allocation." into lmp-dev
|
| 726b6a7540ac997928df1a14dfede872a82f6210 |
11-Aug-2014 |
Ricardo Garcia <rago@google.com> |
Fix to support mono in effect buffer allocation.
Since effects are now enabled for recording, mono cases are possible
and needed to be correctly allocated.
Bug: 16917961
Change-Id: I760fc421f9f90fabb3f0deadd5f9dc61cd284371
ffects.cpp
|
| cd04484f4837b8ca0041d118286ab6a98e84fc75 |
07-Aug-2014 |
Andy Hung <hunga@google.com> |
Extend downsampling ratios greater than 2:1
Also improve robustness to choice of sampling rate or buffer size
such that increasing either by 10x does not cause overflow.
Bug: 12979141
Bug: 15933066
Change-Id: If7989bd745d1bee3bdf811b8b7c978543ccafb65
udioResamplerDyn.cpp
hreads.cpp
|
| 874c4287a4e49c59ac88767751dce00fcd3edb73 |
09-Aug-2014 |
Eric Laurent <elaurent@google.com> |
audio policy: enable more than one sink per audio patch.
Allow creation of audio patches with more than one sink.
More than one sink is enabled when:
- Connecting an input device to output devices on the same audio HW module.
- Connecting an output mix to output devices on the same audio HA module.
All other patches are limited to one sink.
Bug: 16879363.
Change-Id: I95be6948ef29df64e51e5b8ace38c2db7f3e89f2
atchPanel.cpp
|
| 0912a5738d6baf2df7cd62e877240e3807b4b21f |
08-Aug-2014 |
Andy Hung <hunga@google.com> |
Merge "Fix effect and mono sink handling." into lmp-dev
|
| 389cfdbb9a92a438a0d7710321c2964c7ad55eca |
08-Aug-2014 |
Andy Hung <hunga@google.com> |
Fix effect and mono sink handling.
Do not permit mixing to mono sink in AudioFlinger.
Allow effects on mono sink if not Mixer thread (e.g. Record).
Bug: 16863095
Change-Id: I8b232fc1fb3f07bf017020c5d4f9ace644dec6d8
udioFlinger.h
hreads.cpp
|
| 7f04ec2a4ec94ada7ef89db8c50fd1e5c3a93a3b |
07-Aug-2014 |
Eric Laurent <elaurent@google.com> |
Merge "audio policy: add support for USB devices for voice call" into lmp-dev
|
| c2730ba7c5e9559b7499ef5e0d7742deb18c5110 |
21-Jul-2014 |
Eric Laurent <elaurent@google.com> |
audio policy: add support for USB devices for voice call
Add possibility to use sink and source devices not on
the primary HW module for voice calls (e.g. USB headsets).
Bug: 15520724.
Change-Id: Ib27db4ba759b6d91ea1104dc2e35c87733517b30
racks.cpp
|
| 0a0e5f02037f396237461a9840b2fc7271812bdc |
06-Aug-2014 |
Eric Laurent <elaurent@google.com> |
Merge "AudioSystem: add API to query audio HW sync source" into lmp-dev
|
| 7f3f2fa386b8dd917022663bcb01f10ea67061be |
06-Aug-2014 |
Ricardo Garcia <rago@google.com> |
Merge "MediaServer crashing and audio effects not applied to TTS." into lmp-dev
|
| 322bab26dc3fe9bd9c1cbb829dc62ff44f1ae810 |
06-Aug-2014 |
Ricardo Garcia <rago@google.com> |
MediaServer crashing and audio effects not applied to TTS.
A memclear in the EffectChain was out of bounds.
Bug 15432115
Bug 16845751
Change-Id: Ib0adc7fa730d630bb71fb87d51163a26149b16e2
ffects.cpp
|
| 93c3d41bdb15e39dac0faea9c5b60f1637cd477c |
01-Aug-2014 |
Eric Laurent <elaurent@google.com> |
AudioSystem: add API to query audio HW sync source
Add a method to query from the audio HAL the HW sync
source used for a given audio session.
Modify audio policy to select a direct output with HW sync
when requested.
Bug: 16132368.
Change-Id: I03038f9188f2d389f8a5fd76a671854013a4513e
udioFlinger.cpp
udioFlinger.h
|
| cf817a2330936947df94c11859f48771f5596a59 |
05-Aug-2014 |
Eric Laurent <elaurent@google.com> |
audio: fix crashes upon USB device connection
- Audio policy:
Replace unknown device assertion by debug log in setDeviceConnectionState().
It is possible that some kernels indicate connection of a legacy device type
when a USB audio device is connected.
We should just ignore this device.
- Audio flinger:
Make sure FastMixer thread is idle before exiting the normal mixer thread and
closing the output stream.
Change-Id: Ia10a20176a60a8aa56765538093a555fc998508a
hreads.cpp
|
| 7a9b25940309e2967820daa13a7f71e41293d165 |
05-Aug-2014 |
Andy Hung <hunga@google.com> |
Merge "Remove unused TwoTrack process mode from AudioMixer" into lmp-dev
|
| 5241b8cacee86e17223695190959f4752910680f |
05-Aug-2014 |
Andy Hung <hunga@google.com> |
Remove unused TwoTrack process mode from AudioMixer
Legacy code that has not been hooked up.
Change-Id: Ic01a4eddfa1020089d5bc42f58b6a4697698c27a
udioMixer.cpp
udioMixer.h
|
| 44622db93a24219123facbbe14259000f90de728 |
02-Aug-2014 |
Eric Laurent <elaurent@google.com> |
audio flinger: fix service fuzz test crash
Bug: 13744704.
Change-Id: I572c89f88dcab2b8b612ec6860983306f9f159ab
udioFlinger.cpp
|
| e6fa1b7fdc74bec3292d6a10c99f25f9fd989d3f |
16-Jul-2014 |
Hans Boehm <hboehm@google.com> |
Remove 64-bit android_atomic uses from StateQueue.
Use stdatomic.h instead. We're trying to remove android_atomic use
wherever possible. The 64-bit uses seem easiest to remove first.
This cleans up the code, though not as much as C++ <atomic> would,
if it worked everywhere.
(cherry picked from commit f39b560952d3706a7ff47ef0d82c1a836daeea42)
Change-Id: I3c29bdbd5915cb9d47118834a3a742fe296cf87f
tateQueue.cpp
tateQueue.h
|
| 318be3e7d245aea99efa194a16002395b609ab90 |
30-Jul-2014 |
Andy Hung <hunga@google.com> |
Merge "Enable multichannel in AudioFlinger" into lmp-dev
|
| a4daf0b4f934b800a49f199fb8c09409391c8fc0 |
29-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
Fix build warnings
Change-Id: Ife5e40264f43fb3ccb40830228701003fe106bf5
udioMixerOps.h
udioResamplerDyn.cpp
udioResamplerFirProcess.h
ests/resampler_tests.cpp
ests/test-mixer.cpp
ests/test_utils.h
|
| b1ee3954153e9c40280f68d278526ca43c31fdcf |
29-Jul-2014 |
Andy Hung <hunga@google.com> |
Enable multichannel in AudioFlinger
Change-Id: Ibdbdc6ea0b87cdcd15432257a3455b11b3ce14b7
udioFlinger.h
|
| 4dc680607181e6a76f4e91a39366c4f5dfb7b03e |
29-Jul-2014 |
Eric Laurent <elaurent@google.com> |
audio policy: pass session ID to capture activity methods
Indicate the audio session ID when calling getInput(),
startInput(), stopInput(), releaseInput().
Bug: 12378680.
Change-Id: I763793752f93e2f4e1445a5ab217c895af011038
hreads.cpp
racks.cpp
|
| d530d415e60373caf875a37d2d747a51470ee45c |
29-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix rounding in RecordThread resampler buffer computation" into lmp-dev
|
| 8661aaff9e9a4d4c1b57f6a68cdbcab006354ab2 |
28-Jul-2014 |
Andy Hung <hunga@google.com> |
Fix rounding in RecordThread resampler buffer computation
Bug: 16525305
Change-Id: Ie010822b0fa9dec74adde1d1b091ee44c183fc94
hreads.cpp
|
| de3f8392fbf380ba6f09d009b00d7172477389a2 |
28-Jul-2014 |
Eric Laurent <elaurent@google.com> |
rename AudioSystem::newAudioSessionId()
Rename AudioSystem::newAudioSessionId() to
AudioSystem::newAudioUniqueId() as it can be used
also for I/O handles.
Bug: 12378680.
Change-Id: I611ea3b5eb57a4b0774437f477ee87dc4ccc2cc2
udioFlinger.cpp
udioFlinger.h
|
| 03524dcc1215b2c468f5c8c8abe6f1c26239fd71 |
28-Jul-2014 |
Eric Laurent <elaurent@google.com> |
Merge "AudioFlinger: update openInput() and openOutput()" into lmp-dev
|
| cf2c0210c8afbe7d0661ccbbae3835b5ce73c0bf |
26-Jul-2014 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: update openInput() and openOutput()
Add parameters to openInput() and openOutput(): device address,
input source.
Allow caller to specify a given I/O handle
Group parameters in a struct audio_config.
Bug: 12378680.
Change-Id: I7e9af74c0d996561cc13cbee7d9012d2daf33025
udioFlinger.cpp
udioFlinger.h
atchPanel.cpp
|
| 9f81de3452dfb2385bd57dc05456a045174a1ab1 |
28-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
Enable lower latency audio capture on more devices
Bug: 16601366
Change-Id: I573b98631f81b7768f7cc00aa09e560008dea587
hreads.cpp
|
| 9a59276fb465e492138e0576523b54079671e8f4 |
22-Jul-2014 |
Andy Hung <hunga@google.com> |
Add multichannel capability to AudioFlinger
But not enabled (kEnableExtendedChannels == false).
Change-Id: I62f7e31fbd29ad703a9a02f5d1a280b6972dd423
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
udioMixer.h
astMixer.cpp
astMixer.h
hreads.cpp
|
| e93b6b7347a7846c8fd746542364ec11b0cd5124 |
18-Jul-2014 |
Andy Hung <hunga@google.com> |
Add multichannel capability to AudioMixer
tests/test-mixer updated to handle multichannel.
New mixer path is still disabled (kUseNewMixer = false).
Subsequent CL will enable multichannel capability.
Change-Id: I71dafa94cc1d9af567c285e0ded7c5c94e3ff926
udioMixer.cpp
udioMixer.h
udioMixerOps.h
ests/mixer_to_wav_tests.sh
ests/test-mixer.cpp
ests/test_utils.h
|
| 83b8808faad1e91690c64d7007348be8d9ebde73 |
21-Jun-2014 |
Eric Laurent <elaurent@google.com> |
audio flinger: add patch connection between hw modules
Add support for audio device connections between different audio
hw modules.
The patch is performed by creating a bridge between the playback
thread connected to the sink device and the record thread connected
to the source device using a pair of specialized PlaybackTrack and
RecordTrack.
- Added PatchTrack and PatchRecord classes.
- Added TrackBase type to indicate more clearly the track behavior.
- A TrackBase can allocate the buffer or reuse an existing one.
- Factored some code in openOutput() and openInput() for internal use
by PatchPanel.
Bug: 14815883.
Change-Id: Ib9515fcda864610458a4bc81fa8f59096ff4d7db
udioFlinger.cpp
udioFlinger.h
atchPanel.cpp
atchPanel.h
laybackTracks.h
ecordTracks.h
hreads.cpp
hreads.h
rackBase.h
racks.cpp
|
| 17d0842cf33464a79a81b0e78e49514c3419d1cf |
23-Jul-2014 |
Andy Hung <hunga@google.com> |
Merge "Fix multichannel integer resampling and add tests" into lmp-dev
|
| 42b011166ece30969667e0ff9dcf4832568c9c1a |
20-Jul-2014 |
Andy Hung <hunga@google.com> |
Fix multichannel integer resampling and add tests
Change-Id: I384bf8317d4f03616bf9f2b458a8700965d5cf56
udioResamplerFirProcess.h
ests/resampler_tests.cpp
|
| a7e04803ffa0d8d4cc51a122f68bf5038422aabb |
21-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
Remove execute bit
Change-Id: I1cf0c0d9375779b3074af12f4cbf9c1ec07bf4ff
hreads.cpp
|
| 8bce84142802ecdc1794d59488ede629801a5cba |
20-Jul-2014 |
Andy Hung <hunga@google.com> |
Merge "Add RemixBufferProvider to AudioMixer" into lmp-dev
|
| a08810b2feafeec88870c7c1f01efc39ee8e0d78 |
17-Jul-2014 |
Andy Hung <hunga@google.com> |
Add RemixBufferProvider to AudioMixer
This is used to convert between channel masks for input AudioTrack
data.
Change-Id: Iedffaeccb99c7053992152e7b1cb6b2d33b12eef
udioMixer.cpp
udioMixer.h
|
| 34803d594232af5604d893eff7ee40bb57d459a4 |
17-Jul-2014 |
Andy Hung <hunga@google.com> |
Update DownmixerBufferProvider to use CopyBufferProvider
Change-Id: I6b1a8755a1e6c29b916662fd376e9509cda5df55
udioMixer.cpp
udioMixer.h
|
| 1b2fdcb005f9fbe5ad465cec46ec5992b718b5e3 |
17-Jul-2014 |
Andy Hung <hunga@google.com> |
Add CopyBufferProvider class for AudioMixer
AudioMixer::ReformatBufferProvider now uses it as a base class.
Fix ReformatBufferProvider object leak.
Change-Id: If196f844eaaa124a173ffa27afe88098023c2ff9
udioMixer.cpp
udioMixer.h
|
| 4287942b0d6fabe5f99e15cd70c470b04f345de8 |
19-Jul-2014 |
Andy Hung <hunga@google.com> |
Merge "Update DownmixerBufferProvider to use CopyBufferProvider" into lmp-dev
|
| c60780517a882e0905fd36a06b1c45da12e49557 |
19-Jul-2014 |
Andy Hung <hunga@google.com> |
Merge "Add CopyBufferProvider class for AudioMixer" into lmp-dev
|
| 529c61b7e4468a3e21f302f2a92a660249daa722 |
19-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
Fix copy/paste bug
introduced by Change-Id: I4e9b7fa99858b488ac98a441fa70e31dbba1b865
Change-Id: I9a0c8800f5cf3767cc259776f95c96cad972e392
racks.cpp
|
| 91be1fbc424035e70688e60306b755893e0fb0ae |
17-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Move AudioRecord frame count calculations to server" into lmp-dev
|
| 2e8186a258c934798129847f66171df36e95d23e |
17-Jul-2014 |
Andy Hung <hunga@google.com> |
Merge "Fix initial audio glitch on startup" into lmp-dev
|
| 5567aaf4818007cd8e77329683a91c0f5d7a8837 |
17-Jul-2014 |
Andy Hung <hunga@google.com> |
Fix initial audio glitch on startup
Also happens when restarting AudioFlinger.
Bug: 16322824
Change-Id: I382b20174cb0639e04316130141f2094d7dd0cb3
hreads.cpp
|
| 7410591dad836434c72ddee66680802708b70c10 |
03-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
Move AudioRecord frame count calculations to server
Buffer frame count and notification frame count
are now calculated by server instead of by client.
The server has more information and can do a better job.
Also fix a few bugs:
- If a fast track was re-created, even with same pipe depth, it would fail.
Now it can correctly re-create a fast track provided the pipe depth is same.
- Notification frame count for fast tracks was calculated by client
as 1/2 of the total frame count, which is a large value due to the pipe.
Now the notification frame count is set by server to the HAL frame count.
This should reduce latency for fast tracks.
- EVENT_OVERRUN were happening frequently when there was sample rate conversion,
because the client didn't know about the sample rate conversion,
and under-estimated the necessary buffer size. Now since server
calculates the buffer sizes, EVENT_OVERRUN is unlikely.
- RecordThread::createRecordTrack_l was checking for mono and stereo
for fast tracks. This is not necessary, and now we can handle a
multi-channel fast track.
Bug: 7498763
Change-Id: I0c581618e8db33084d5ff9ed50a592990c9749e8
hreads.cpp
|
| ec40d284218466d8f0e832e7eb88e6ea6c479c88 |
16-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
Add audio_input_flags_t to IAudioFlinger::openInput
For backward compatibility, until flags are correctly calculated,
we will assume that the request is for a low latency input stream.
Change-Id: I76746834e870df00833dc77cbdaa2edd2ffeec95
udioFlinger.cpp
udioFlinger.h
|
| 6e6704c06d61bc356e30c164081e5bcffb37920c |
03-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
Improve dumpsys for capture threads
Show per-track sample rate, and whether there is a fast capture thread
Change-Id: If6a08a40d1eeba0690aa8da3c541f845ec101b7f
hreads.cpp
racks.cpp
|
| da3bd9f12883f8344d2db00ce2fa47511ed9bef6 |
16-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
Add audio_input_flags_t to HAL open_input_stream
For backward compatibility, until flags are added to openInput(),
we will assume that the request is for a low latency input stream.
Change-Id: I5e6f7e574418917ff1b41b0a47199203c69cb40f
udioFlinger.cpp
|
| cd704219d22ec51660103684a680caf2c3a12dad |
15-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
const correctness on downmix and upmix
Change-Id: I0f51470a6d58a22310ce7cda80afc55e9d6df1dc
hreads.cpp
|
| 2829edccd7d2bb8244246f316face82b650b8949 |
18-Jun-2014 |
aarti jadhav-gaikwad <agaikwad@nvidia.com> |
stagefright: Pass audio aac sub formats in offloadinfo
Pass audio aac sub formats in offloadinfo according to
aac profile. Audio HAL can take decision about offload
using DSP capabilities
Change-Id: If269a3654b5d2b09c183212b0646ef03e06f2d8f
udioFlinger.cpp
|
| 463be250de73907965faa6a216c00312bf81e049 |
11-Jul-2014 |
Andy Hung <hunga@google.com> |
Return negotiated format with HAL in AudioFlinger
Fixes an issue caused by extended precision and the FastMixer
when the format to the FastMixer was returned instead of the
HAL format.
Change-Id: I2be9d55d5c87d82e46b5200b07fcdd986ebde08a
hreads.cpp
hreads.h
|
| 0dba9b1719a598f5d78cbcef4b0b9e47313f1037 |
09-Jul-2014 |
Andy Hung <hunga@google.com> |
Merge "Add floating point volume handling to AudioMixer"
|
| 91848a568e6ef4d6b214b81d3f82b6d49d00ffb6 |
09-Jul-2014 |
Andy Hung <hunga@google.com> |
Merge "Enable extended precision PCM output in AudioFlinger"
|
| 5e58b0abe5b6c8f5bd96a8f78bbeeeb4d3892020 |
24-Jun-2014 |
Andy Hung <hunga@google.com> |
Add floating point volume handling to AudioMixer
Use floating point volume in AudioMixer mixing when floating
point input is used with the new mixer engine.
AudioResampler is updated to take floating point volume to match.
Both legacy integer and floating point mixer engines work.
For now, integer volume is used when the new mixer engine
runs in integer input mode, for backward compatibility with
the legacy mixer. The new mixer engine will generally run in
floating point input mode. When the legacy path is removed,
the integer volumes will be removed.
Change-Id: I79e80c292ae7c8b8bdd0aa371a1b2c3a1b618290
ndroid.mk
udioMixer.cpp
udioMixer.h
udioMixerOps.h
udioResampler.cpp
udioResampler.h
udioResamplerDyn.cpp
udioResamplerDyn.h
udioResamplerSinc.cpp
udioResamplerSinc.h
racks.cpp
est-resample.cpp
ests/resampler_tests.cpp
|
| a494e82c3c73508b4d3cfe89e9134de94e12fd31 |
09-Jul-2014 |
Andy Hung <hunga@google.com> |
Enable extended precision PCM output in AudioFlinger
Change-Id: I7c0907c7b2369681975d8ea0192b722d7ed7a867
udioFlinger.h
|
| 3d638b30dcc67ef9eaa389160ef285cda055ed64 |
08-Jul-2014 |
Andy Hung <hunga@google.com> |
Merge "Replace int bitDepth with audio_format_t in Resampler"
|
| 3348e36c51e91e78020bcc6578eda83d97c31bec |
07-Jul-2014 |
Andy Hung <hunga@google.com> |
Replace int bitDepth with audio_format_t in Resampler
Remove mBitDepth from class (not used).
Replace with audio_format_t in factory method to distinguish
between float and pcm 16-bit.
Change-Id: I166860796c68285077ef4458d8758d19b82523f9
udioMixer.cpp
udioResampler.cpp
udioResampler.h
udioResamplerCubic.h
udioResamplerDyn.cpp
udioResamplerDyn.h
udioResamplerSinc.cpp
udioResamplerSinc.h
racks.cpp
est-resample.cpp
ests/resampler_tests.cpp
|
| 075abae2a954bf3edf18ad1705c2c0f188454ae0 |
10-Apr-2014 |
Andy Hung <hunga@google.com> |
Add and enable multichannel for audio resampler
Change-Id: I2b86fb73d70abc4c456f7567270a888086b301d4
Signed-off-by: Andy Hung <hunga@google.com>
udioResampler.cpp
udioResamplerDyn.cpp
udioResamplerDyn.h
ests/resampler_tests.cpp
|
| 68ffa200de7c4662c088851a328923be715c6c24 |
10-Apr-2014 |
Andy Hung <hunga@google.com> |
Add multichannel to audio resample processing
Change-Id: If6bfbf74000520713f9ba19488a475fc2eefc271
Signed-off-by: Andy Hung <hunga@google.com>
udioResamplerFirProcess.h
|
| 5c68f959eaa2e02fed5643c78e281fff42bcc0a2 |
07-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "IAudioFlinger::openRecord now suggests notificationFrames"
|
| 7df8c0b799d8f52d6386e03313286dbd7d5cdc7c |
03-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
IAudioFlinger::openRecord now suggests notificationFrames
Change-Id: I08885cc381d03c522a23289e74f0e1ed46563863
udioFlinger.cpp
udioFlinger.h
hreads.cpp
hreads.h
|
| 665470b36f202bcc8ee2f7417f68fd2608dd07c1 |
04-Jul-2014 |
Eric Laurent <elaurent@google.com> |
audio: deprecate audio_stream_frame_size()
Bug: 15000850.
Change-Id: I955a6efa620b8f70fa5193c7853cbe88e879c9c3
hreads.cpp
|
| 5ba4440c11eb975ec0e104e0af1981838f42f57c |
04-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove obsolete IAudioFlinger::channelCount()"
|
| 296b741e8eb38e749e3202182f703a2e30ee5f1f |
18-Jun-2014 |
Andy Hung <hunga@google.com> |
Add new AudioMixer processing hooks
This change adds new process and thread hooks for
multi-format handling.
It is enabled by setting kUseNewMixer = true.
Change-Id: I262a3d2e4411f8cef7370a497b77a34eb55f1f86
udioMixer.cpp
udioMixer.h
udioMixerOps.h
|
| f92f22becdf7fce1f55d5ebd80ac2caa2ad55602 |
30-Jun-2014 |
Andy Hung <hunga@google.com> |
Merge "Add enabling variable for extended precision audio"
|
| 6146c08f0c3dd8b9e5788063aa433f304a810602 |
18-Mar-2014 |
Andy Hung <hunga@google.com> |
Add enabling variable for extended precision audio
Set AudioFlinger::kEnableExtendedPrecision = true to enable
extended precision. Enabling will be required for devices (such as
USB) which report 24 bit or 32 bit sink formats.
Change-Id: I0dc1d7a4f7607086d7b536ea0e43aef0e696f2ee
udioFlinger.cpp
udioFlinger.h
hreads.cpp
|
| 4d23ca370dd0ce584f49a80ef9dfcdbb75ba2c8e |
13-May-2014 |
Glenn Kasten <gkasten@google.com> |
Use sp<> instead of raw pointer for FastMixer
This avoids worry about the thread lifetime during tear-down.
Bug: 15858655
Change-Id: I896d24317bd6f53a574645cdf4cd514eaa205dd5
hreads.cpp
hreads.h
|
| eaf07697a9146d27ca863db5c43da64af4f8c94a |
23-Jun-2014 |
Andy Hung <hunga@google.com> |
Merge "Add test-mixer AudioMixer test program"
|
| e4fc42359cdd9786e521054a3a0491d6bc3a9e1c |
18-Jun-2014 |
Andy Hung <hunga@google.com> |
Add test-mixer AudioMixer test program
The shell script mixer_to_wav_tests.sh shows how to use
test-mixer.
Change-Id: Ia7f1a368972c9c33fadc96df4cb1fc8b22446c8c
ests/Android.mk
ests/mixer_to_wav_tests.sh
ests/test-mixer.cpp
|
| 9b3cd171710991625a94dea645248f708cb957fb |
20-Jun-2014 |
Andy Hung <hunga@google.com> |
Merge "Extract out test utility functions from resampler_tests"
|
| ab5cdbaf65ca509681d2726aacdf3ac8bfb6b3fa |
10-Jun-2014 |
Eric Laurent <elaurent@google.com> |
AudioTrack: add support for compressed audio
Add support for compressed audio playback by use
of an AudioTrack attached to a direct output thread.
Bug: 9428304.
Change-Id: I4a61be9cf0e31003ca85935d6e0ee38ca8192e03
laybackTracks.h
hreads.cpp
racks.cpp
|
| c0e5ec8e2d8db15b97094374d0a248e041304b62 |
17-Jun-2014 |
Andy Hung <hunga@google.com> |
Extract out test utility functions from resampler_tests
These will be used for mixer tests later.
Change-Id: I22b1c89857058cfb2450afe15e80d6c9306a31f1
ests/Android.mk
ests/resampler_tests.cpp
ests/test_utils.h
|
| 5094675aa55458414e8b00a86a104a32ea1d55c0 |
16-Jun-2014 |
Glenn Kasten <gkasten@google.com> |
am a3603693: Merge "Fix deadlock" into lmp-preview-dev
* commit 'a3603693fb784d7e1a820ee4629a69730a0bb424':
Fix deadlock
|
| a3603693fb784d7e1a820ee4629a69730a0bb424 |
16-Jun-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix deadlock" into lmp-preview-dev
|
| 3d61bc1ffc8afc8d7be3b0d4205c9b5ba6daf2e8 |
16-Jun-2014 |
Glenn Kasten <gkasten@google.com> |
Fix deadlock
Bug: 15591052
Change-Id: I76154e1640646d2ae8fae65e163e1f3ed1be17dc
hreads.cpp
|
| 3bcffa136909c1fb6e88ee4efd12ccac18360a85 |
13-Jun-2014 |
Eric Laurent <elaurent@google.com> |
force command flush after releasing a reference to shared memory
After releasing a reference to a shared memory interface,
IPCThreadState::self()->flushCommands() must be called to ensure
that the corresponding reference is also cleared immediately in the
remote process. Otherwise the binder implementaiton will not
push the clear systematically causing the memory region not to be freed.
See issues 10711502 and 2801375.
Bug: 14057166.
Change-Id: If55f36f00d452d6cf00cf83bd2fba1c8f3abcb57
racks.cpp
|
| 20d16a8af2dc95e827be693edabde4f09eb90c32 |
13-Jun-2014 |
Eric Laurent <elaurent@google.com> |
am 3bcffa13: force command flush after releasing a reference to shared memory
* commit '3bcffa136909c1fb6e88ee4efd12ccac18360a85':
force command flush after releasing a reference to shared memory
|
| b6be7f22a82ee3bad8bcc709d21e72fc4727da09 |
12-Jun-2014 |
Bryant Liu <bryant_liu@htc.com> |
frameworks: av: fix bug in the volume management
which can cause volume levels to be applied twice. After applying volume
in the effect chain, the remaining volume needs to be converted back to
floating point before providing to mAudioMixer.
Bug: 15583440
Change-Id: I3c884f95b0fa9ad1b554285b069601c850c54776
Signed-off-by: Glenn Kasten <gkasten@google.com>
hreads.cpp
|
| 14c403fbab456a4a63586a767fbae498f5f8c260 |
12-Jun-2014 |
Glenn Kasten <gkasten@google.com> |
am 0ac53364: Merge "Fix non-monotonic AudioTrack::getTimestamp after pause" into lmp-preview-dev
* commit '0ac53364b79dc72ee0c09f0efc46a22ab4374dbf':
Fix non-monotonic AudioTrack::getTimestamp after pause
|
| 0ac53364b79dc72ee0c09f0efc46a22ab4374dbf |
11-Jun-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix non-monotonic AudioTrack::getTimestamp after pause" into lmp-preview-dev
|
| ced6e74215937182fe2f9f6b0867f7c28ccd02c1 |
10-Jun-2014 |
Glenn Kasten <gkasten@google.com> |
Fix non-monotonic AudioTrack::getTimestamp after pause
Bug: 15523502
Change-Id: Ifd4aa7fca197bc041c1620fc3f7d953a8902551a
laybackTracks.h
racks.cpp
|
| 54188de28dc7766ba578a7ce01c61151c6febdc0 |
03-Jun-2014 |
Andy Hung <hunga@google.com> |
Update resampler to fetch exactly the frames needed
This avoids unnecessary overfetching/releasing, which
could cause problems with the upstream AudioBufferProvider.
A one input sample delay is placed on the stream compared
with the previous implementation.
This change only affects the Dynamic resampler.
Bug: 15548547
Change-Id: Ic7fcff130e0081b4724cfb5a00dc8e8b4a8b1af3
(cherry picked from commit 717007429a50c02d2acc704a8c1ebbe6760a2c22)
udioResamplerDyn.cpp
|
| d8754e07e23eecb68068c0c392aeb0428c529e4e |
10-Jun-2014 |
Eric Laurent <elaurent@google.com> |
DO NOT MERGE - AudioFlinger: increase per client memory pool
Temporary workaround for issue 14057166 allowing
AudioTracks from a given client to use up to 4MB
of shared memory.
Bug: 14057166.
Change-Id: I19adb8d424ed170df61c1cd706be3d10788edfc1
udioFlinger.cpp
|
| 51fabcd581f293db52997b0d43b2ef606d5759ff |
09-Jun-2014 |
Andy Hung <hunga@google.com> |
Merge "Add tests directory for AudioFlinger"
|
| 32ff60c6ec297256b59b6ce4f3cb341a0f4638dd |
06-Jun-2014 |
Eric Laurent <elaurent@google.com> |
am 45df31f5: Merge "AudioFlinger: fix PatchPanel debug log." into lmp-preview-dev
* commit '45df31f511ba9ae9b45c769fc546eca4bcf4bbdb':
AudioFlinger: fix PatchPanel debug log.
|
| 45df31f511ba9ae9b45c769fc546eca4bcf4bbdb |
05-Jun-2014 |
Eric Laurent <elaurent@google.com> |
Merge "AudioFlinger: fix PatchPanel debug log." into lmp-preview-dev
|
| c263ca0ad8b6bdf5b0693996bc5f2f5916e0cd49 |
05-Jun-2014 |
Glenn Kasten <gkasten@google.com> |
Squashed commit of the following:
commit 9128d6ffec43731d723f9b394f243d940f4c7e41
Author: Glenn Kasten <gkasten@google.com>
Date: Tue May 13 10:38:42 2014 -0700
Use of fast capture by normal capture
Will only configure fast capture path if the input buffer size is less than
10 ms and the input sample rate is same as the primary output sample rate.
Change-Id: I4a7cdc6069d750845412c626d27e83f72a1ab397
commit 2e5e0806a5abe7499848358ef5fde5c26405000d
Author: Glenn Kasten <gkasten@google.com>
Date: Mon Jun 2 08:29:22 2014 -0700
Add mPrimaryOutputSampleRate
Change-Id: I46b527fc3f2b5a5720a74b4f0b9a8f2e0d570b09
commit baf1d73467923996d1b1f2a9237260cc5697e050
Author: Andy Hung <hunga@google.com>
Date: Fri May 30 10:42:03 2014 -0700
Change parameter type for volume to float in AudioMixer
Change-Id: I4da1505ce852505f86f8e5b87f60e8edceeb30e0
commit 40fe20fa9760cd03c69778c2021cf7a490d75ece
Author: Andy Hung <hunga@google.com>
Date: Fri May 30 10:35:47 2014 -0700
Rename UNITY_GAIN to UNITY_GAIN_INT in AudioMixer
Change-Id: Ic040311305026f0b4c4280a5b3bef7a447ac1da3
commit 37c9a2b49f876abc5ff537a9ec036d7f0a423775
Author: Andy Hung <hunga@google.com>
Date: Thu May 29 21:33:13 2014 -0700
Refactor setVolumeRampVariables in AudioMixer
Change-Id: I8fcf3101bcea292de7c65433fa578f1c9cdd0974
commit 397070eca31f121d5d3993de1bfea99aaea5d4f3
Author: Andy Hung <hunga@google.com>
Date: Thu May 29 18:52:38 2014 -0700
Fix floating point output from mixer
A buffer pointer was being erroneously reset to buffer start,
potentially causing an audio glitch. The floating point
output mode is not enabled at this time, but will be in
the future.
Change-Id: If8b6414d232f064f3a2e2c5a6da889a91b27fb24
commit 2e61aa5b33b2247bbc5d4eaa0b519df9accd4bbc
Author: Andy Hung <hunga@google.com>
Date: Fri May 23 21:22:17 2014 -0700
Add multiple format capability to FastMixer
Floating point data from MixerThread into FastMixer.
Multiple output format capability from FastMixer to Sink.
Change-Id: I0da17810ee71381a39a006c46faec71108d22c26
commit b9ea653c702a785bbd23a66c5e588d40b4192c4e
Author: Andy Hung <hunga@google.com>
Date: Thu May 29 15:53:09 2014 -0700
Avoid resetting BufferProviders in mixer unnecessarily
Change-Id: Iad85c4dfd21be1dbf89dc11906106b34219376f8
commit 7f1a6d6da21c616f80cf9ba21bea11b419ec561b
Author: Andy Hung <hunga@google.com>
Date: Tue May 27 12:32:17 2014 -0700
Update dynamic resampler buffer fetching
Make the criteria tight for fetching to avoid storing
excessive frame data internal to the resampler.
This should reduce jitter in frame delivery computation.
Bug: 14962343
Change-Id: I7adaf714d11c272696ccdbf218bda994c7217477
commit b5e4aac07b9a02f0c803c090058602b03ac09ebb
Author: Glenn Kasten <gkasten@google.com>
Date: Tue May 27 12:30:54 2014 -0700
Allow kFastTrackMultiplier to be specified per device
Change-Id: I4eaaaf038df720cec4f5d9221d1b632970f9e3dd
commit b93cd97a52af31122df2da2cc0415cda888c8c73
Author: Andy Hung <hunga@google.com>
Date: Fri May 23 21:13:31 2014 -0700
Rename mixBuffer to mMixerBuffer in FastMixer
Likewise mixBufferState becomes mMixerBufferState.
This harmonizes with the naming in AF::MixerThread.
Change-Id: I1255d7c07cc2c6ee925d7430925236d2bd163122
commit 8340758622b9711365a8801806cbdf934803c63f
Author: Andy Hung <hunga@google.com>
Date: Mon May 12 16:51:41 2014 -0700
Add multiple format capability to AudioMixer
Change-Id: I04ac1cafd90b6ed652f8d51888ad07576678f0bc
Signed-off-by: Andy Hung <hunga@google.com>
commit 6b695b9d094820c232a897a3fabbe83d2b7193fe
Author: Glenn Kasten <gkasten@google.com>
Date: Thu Mar 13 14:59:31 2014 -0700
Start adding FastCapture based on FastThread WIP
This version supports at most one fast capture client.
Change-Id: Idf609bfc80ae22433433d66a5232c043c65506df
commit e951ad05a2c388471d7e2806d91e7d51325a150a
Author: Glenn Kasten <gkasten@google.com>
Date: Mon May 12 11:06:26 2014 -0700
Move validation of frameCount from set to openRecord_l
This move is needed because frameCount is validated on server side for
fast tracks (as should be done for normal tracks too).
Change-Id: I6d99e80869fd90fab373cf60ef348c01f075fbca
commit 73e76992dbba794894837c38e5472312ea829cf3
Author: Glenn Kasten <gkasten@google.com>
Date: Tue May 13 10:41:52 2014 -0700
Allow track buffer "allocation" to be from pipe
Change-Id: Ib9ac170f8e8b7746b3588157a56cbee3b753a1cb
commit 60de1d7ded05c6304037d4858b401094b1d2b4d3
Author: Andy Hung <hunga@google.com>
Date: Fri May 9 15:02:21 2014 -0700
Add format parameter to getTrackName() and track_t
Change-Id: Ia152a839014e235fbfb656104c15d7c1b456d02e
Signed-off-by: Andy Hung <hunga@google.com>
Change-Id: Ied0ade8b25d23e89bb03319a7e3135c238f735b9
ndroid.mk
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
udioMixer.h
udioResamplerDyn.cpp
astCapture.cpp
astCapture.h
astCaptureState.cpp
astCaptureState.h
astMixer.cpp
astMixer.h
astMixerState.cpp
astMixerState.h
tateQueueInstantiations.cpp
hreads.cpp
hreads.h
rackBase.h
racks.cpp
|
| 24478d47fc631ab33208f4dd9d034abb6839c992 |
05-Jun-2014 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: fix PatchPanel debug log.
Bug: 14815883.
Change-Id: I71cf2ef74a96d3cdb27f2beb1b76b8979c738335
atchPanel.cpp
|
| f1caa55fbe2a32a27968d7be2f0cd134e412db4d |
05-Jun-2014 |
Andy Hung <hunga@google.com> |
Merge "Update resampler to fetch exactly the frames needed"
|
| aaaea95d23b78313b7ddf1f8f670c9b0110474ed |
04-Jun-2014 |
Eric Laurent <elaurent@google.com> |
am 83438ce6: Merge "audio policy: add permission for routing APIs." into lmp-preview-dev
* commit '83438ce6cbbff49ca26d4bd694ffdad4842ded5a':
audio policy: add permission for routing APIs.
|
| 717007429a50c02d2acc704a8c1ebbe6760a2c22 |
03-Jun-2014 |
Andy Hung <hunga@google.com> |
Update resampler to fetch exactly the frames needed
This avoids unnecessary overfetching/releasing, which
could cause problems with the upstream AudioBufferProvider.
A one input sample delay is placed on the stream compared
with the previous implementation.
This change only affects the Dynamic resampler.
Change-Id: Ic7fcff130e0081b4724cfb5a00dc8e8b4a8b1af3
udioResamplerDyn.cpp
|
| 8f415570afb174f0781d8bf6b1c0fa6bdecca7cd |
03-Jun-2014 |
Andy Hung <hunga@google.com> |
Merge "Change parameter type for volume to float in AudioMixer"
|
| 93f87fe162f6c18a296451d388ea68e76fd67fb6 |
03-Jun-2014 |
Andy Hung <hunga@google.com> |
Merge "Rename UNITY_GAIN to UNITY_GAIN_INT in AudioMixer"
|
| 9dc6699fd613c29fe08ef7bc0d32052c8ce297c1 |
03-Jun-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Use of fast capture by normal capture"
|
| 740a5ca81f6498b34baea04bb0fb3fc29fe1e135 |
03-Jun-2014 |
Andy Hung <hunga@google.com> |
Merge "Refactor setVolumeRampVariables in AudioMixer"
|
| 6dbb5e3336cfff1ad51d429fcb847307c06efd61 |
13-May-2014 |
Glenn Kasten <gkasten@google.com> |
Use of fast capture by normal capture
Will only configure fast capture path if the input buffer size is less than
10 ms and the input sample rate is same as the primary output sample rate.
Change-Id: I4a7cdc6069d750845412c626d27e83f72a1ab397
udioFlinger.h
hreads.cpp
hreads.h
racks.cpp
|
| 4ea00a25cf85877b48ebd1e15a657cfaab29af58 |
02-Jun-2014 |
Glenn Kasten <gkasten@google.com> |
Add mPrimaryOutputSampleRate
Change-Id: I46b527fc3f2b5a5720a74b4f0b9a8f2e0d570b09
udioFlinger.cpp
udioFlinger.h
|
| 5284ed53c14cd4d15bd793000ede7166d143e69a |
29-May-2014 |
Eric Laurent <elaurent@google.com> |
audio policy: add permission for routing APIs.
Add check for signature or system permission
"android.permission.MODIFY_AUDIO_ROUTING" for routing
APIs.
Bug: 14815883.
Change-Id: Iefa0b9d90127a53e03a2e6d50dc93d8f4ae63e2f
erviceUtilities.cpp
erviceUtilities.h
|
| 43466a133984182b250453cea244eb146e83c010 |
02-Jun-2014 |
Eric Laurent <elaurent@google.com> |
am 62aaabb3: audio policy: fix unitialized hw module in audio port
* commit '62aaabb3905c61bb7acd6037414c206240a31c32':
audio policy: fix unitialized hw module in audio port
|
| 62aaabb3905c61bb7acd6037414c206240a31c32 |
02-Jun-2014 |
Eric Laurent <elaurent@google.com> |
audio policy: fix unitialized hw module in audio port
The hw module handle was not initialized when creating
and audio port configuration from an input stream descriptor.
Bug 15376641.
Change-Id: I6c784cded68a5c665fd22c3643765c63f70eb241
atchPanel.cpp
|
| d28148537693b19320f3c3b5b34816d45542e9bd |
02-Jun-2014 |
Andy Hung <hunga@google.com> |
Merge "Fix floating point output from mixer"
|
| f947dbce4390f2c3c460325d37002a34f09c0b74 |
01-Jun-2014 |
Glenn Kasten <gkasten@google.com> |
Remove obsolete IAudioFlinger::channelCount()
Change-Id: Ie623edae2e795f9155f1f452fe4e6c7217a4a4c8
udioFlinger.cpp
udioFlinger.h
hreads.h
|
| e28456465fca86ed1b8b2225de073cfba64c7c42 |
01-Jun-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Start adding FastCapture based on FastThread WIP"
|
| 481c8837c7f67852e06137d636915ea6e96247e3 |
31-May-2014 |
Eric Laurent <elaurent@google.com> |
am e1715a46: audio policy: add audio port gain control
* commit 'e1715a465a29db625da9d0ea365edf371e39e201':
audio policy: add audio port gain control
|
| e1715a465a29db625da9d0ea365edf371e39e201 |
20-May-2014 |
Eric Laurent <elaurent@google.com> |
audio policy: add audio port gain control
Implement setAudioPortConfig() API to configure an
audio port.
Currently limited to gain control.
Bug: 14815883.
Change-Id: Ic1b268e5ba5c277e9f5b8fa63c81dd7c0b250024
atchPanel.cpp
|
| 6be494077f8d7970f3a88129c5d139c5a0c88f6d |
30-May-2014 |
Andy Hung <hunga@google.com> |
Change parameter type for volume to float in AudioMixer
Change-Id: I4da1505ce852505f86f8e5b87f60e8edceeb30e0
udioMixer.cpp
udioMixer.h
astMixer.cpp
hreads.cpp
|
| 97ae82408469286426d8dc57a361350f74e24a8b |
30-May-2014 |
Andy Hung <hunga@google.com> |
Rename UNITY_GAIN to UNITY_GAIN_INT in AudioMixer
Change-Id: Ic040311305026f0b4c4280a5b3bef7a447ac1da3
udioMixer.cpp
udioMixer.h
racks.cpp
|
| 5866a3b9689035580c31f1a02037b8176d7508a7 |
30-May-2014 |
Andy Hung <hunga@google.com> |
Refactor setVolumeRampVariables in AudioMixer
Change-Id: I8fcf3101bcea292de7c65433fa578f1c9cdd0974
udioMixer.cpp
|
| f8a106ae5c40735de6c75e7c12d5e0e2e22f12cf |
30-May-2014 |
Andy Hung <hunga@google.com> |
Fix floating point output from mixer
A buffer pointer was being erroneously reset to buffer start,
potentially causing an audio glitch. The floating point
output mode is not enabled at this time, but will be in
the future.
Change-Id: If8b6414d232f064f3a2e2c5a6da889a91b27fb24
udioMixer.cpp
|
| 671714b80cad4725d07691fbd8284cb869a0c27f |
30-May-2014 |
Eric Laurent <elaurent@google.com> |
am 6a94d69d: audio policy: implement routing control
* commit '6a94d69dc4f32abb53c466a96f905bb199be6417':
audio policy: implement routing control
|
| 47aaf6da5d7ecc2ad28de004d1820b64b02dd344 |
30-May-2014 |
Andy Hung <hunga@google.com> |
Merge "Add multiple format capability to FastMixer"
|
| 1258c1ab592a899fabb1e31eb5db2ef413b6f38a |
24-May-2014 |
Andy Hung <hunga@google.com> |
Add multiple format capability to FastMixer
Floating point data from MixerThread into FastMixer.
Multiple output format capability from FastMixer to Sink.
Change-Id: I0da17810ee71381a39a006c46faec71108d22c26
astMixer.cpp
astMixer.h
hreads.cpp
|
| 6a94d69dc4f32abb53c466a96f905bb199be6417 |
20-May-2014 |
Eric Laurent <elaurent@google.com> |
audio policy: implement routing control
Add implementation of audio routing control via AudioSystem APIs.
The following APIs are implemented:
- listAudioPorts(): return a list of devices and output/input mixers ports
that can be used as sources or sinks for audio patches.
- createAudioPatch()/releaseAudioPatch(): create/release a connection patch between
two audio ports (e.g. to connect input from an HDMI device to a speaker output device).
Only one client application can own a patch from a given source.
When an audio port (device or mix) is part of an application created patch, its routing cannot
not be changed by a policy decision.
- listAudioPatches(): return a list of existing patches.
Each audio port addition/removal and each audio patch creation/release increments a generation count.
This generation count is used to ensure consistency betwen calls to
listAudioPorts() and listAudioPatches().
Bug: 14815883.
Change-Id: I022b638c2f5f0bb41543c7cfca7488fb45cfdd80
atchPanel.cpp
|
| 1d26ddf715182d6082804ff70086c47fb8c354b8 |
30-May-2014 |
Andy Hung <hunga@google.com> |
Avoid resetting BufferProviders in mixer unnecessarily
Change-Id: Iad85c4dfd21be1dbf89dc11906106b34219376f8
udioMixer.cpp
|
| 7dac33136d71ea566de4531235f5fbbd28048f0c |
29-May-2014 |
Andy Hung <hunga@google.com> |
Merge "Update dynamic resampler buffer fetching"
|
| 951f455566775e5f01e67c5ee26863d7d19209d7 |
20-May-2014 |
Eric Laurent <elaurent@google.com> |
DO NOT MERGE - audioflinger: first patch panel implementation.
Added a new PatchPanel subclass to AudioFlinger
to handle audio ports and audio patches configuration
and connection.
The first implementation does not add new functionnality.
AudioPolicyManager uses patch panel interface to control
device routing.
AudioFlinger:
- Added PatchPanel class. The first implementation does not
add new functionnality. PatchPanel handles routing commands
for audio HAL after 3.0 or converts to setParameters for audio
HALs before 3.0.
- Added config events to ThreadBase to control synchronized
audio patch connection.
AudioPolicyManager:
- Use PatchPanel API to control device selection isntead of setParameters.
- New base class AudioPort common to audio device descriptors
and input output stream profiles. This class is RefBase and groups
attributes common to audio ports.
- Use same device selection flow for input as for outputs:
getNewInputDevice -> getDeviceForInptusiource -> setInputDevice
Change-Id: Idaa5a883b19a45816651c58cac697640dc717cd9
ndroid.mk
udioFlinger.cpp
udioFlinger.h
atchPanel.cpp
atchPanel.h
hreads.cpp
hreads.h
|
| 1cda6afaf6207a41303e653a6ecd7909d73186eb |
11-Apr-2014 |
Eric Laurent <elaurent@google.com> |
DO NOT MERGE - IAudioFlinger interface extension for patch panel
Change-Id: Iaabe0a7e315d5725e00a74a6ed40339b98f20330
udioFlinger.h
|
| 411cb8e45442d16d983a38722c7c7bce487bde6b |
27-May-2014 |
Andy Hung <hunga@google.com> |
Update dynamic resampler buffer fetching
Make the criteria tight for fetching to avoid storing
excessive frame data internal to the resampler.
This should reduce jitter in frame delivery computation.
Bug: 14962343
Change-Id: I7adaf714d11c272696ccdbf218bda994c7217477
udioResamplerDyn.cpp
|
| fe9adc4850572cb04a9cf7877d0e77d0c1fde84e |
27-May-2014 |
Eric Laurent <elaurent@google.com> |
DO NOT MERGE - audioflinger: fix deadlock upon AudioRecord creation error
AudioFlinger:openRecord() should not hold mClientLock when
releasing the local reference on AudioRecord as the destructor will
also lock mClientLock.
Same fix for AudioFlinger::createTrack().
Also make sure that AudioFlinger::createEffect() holds mClientLock
when clearing local reference to the Client in case of error.
Regression introduced by 021cf9634ab09c0753a40b7c9ef4ba603be5c3da
Bug: 15118096.
Change-Id: Ie961c398c8e0460bca9b95e2ee4ce6859316c275
udioFlinger.cpp
|
| d6461906586032716fc5ff9970c8c6cacf235a20 |
22-May-2014 |
Eric Laurent <elaurent@google.com> |
DO NOT MERGE - audioflinger: test min HAL version instead of current
Change-Id: Ibce2353b4390329b4c1a453e829178e3986c25a7
udioFlinger.cpp
|
| 546734b3ab577d46afe863515104a062e88a109b |
02-Apr-2014 |
Andy Hung <hunga@google.com> |
Add tests directory for AudioFlinger
Native tests for audio resampler through gtest framework.
Change-Id: Ia340ca51f9b9b78d67e03e3541e4462b98578c95
ests/Android.mk
ests/build_and_run_all_unit_tests.sh
ests/resampler_tests.cpp
ests/run_all_unit_tests.sh
|
| ecc03733bfd3262ffadef3166e6be23b539c505c |
28-May-2014 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: first patch panel implementation."
|
| 5e49780692b4f23966be89ac32b19c3c1ead4e85 |
28-May-2014 |
Eric Laurent <elaurent@google.com> |
Merge "IAudioFlinger interface extension for patch panel"
|
| c9cd62bf52dd5ea8dc292fb7bc80ff51ed1a79bb |
28-May-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Allow kFastTrackMultiplier to be specified per device"
|
| 0349009fd19f89f8414c428f6b71b369f7546085 |
27-May-2014 |
Glenn Kasten <gkasten@google.com> |
Allow kFastTrackMultiplier to be specified per device
Change-Id: I4eaaaf038df720cec4f5d9221d1b632970f9e3dd
hreads.cpp
|
| b9e722bdf00333557578497c5abfc77c5d779ecb |
28-May-2014 |
Andy Hung <hunga@google.com> |
Merge "Rename mixBuffer to mMixerBuffer in FastMixer"
|
| 018d324dc599c5f4238a4cd631baac47c28d59fe |
27-May-2014 |
Andy Hung <hunga@google.com> |
Merge "Add multiple format capability to AudioMixer"
|
| 1c333e252cbca3337c1bedbc57a005f3b7d23fdb |
20-May-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: first patch panel implementation.
Added a new PatchPanel subclass to AudioFlinger
to handle audio ports and audio patches configuration
and connection.
The first implementation does not add new functionnality.
AudioPolicyManager uses patch panel interface to control
device routing.
AudioFlinger:
- Added PatchPanel class. The first implementation does not
add new functionnality. PatchPanel handles routing commands
for audio HAL after 3.0 or converts to setParameters for audio
HALs before 3.0.
- Added config events to ThreadBase to control synchronized
audio patch connection.
AudioPolicyManager:
- Use PatchPanel API to control device selection isntead of setParameters.
- New base class AudioPort common to audio device descriptors
and input output stream profiles. This class is RefBase and groups
attributes common to audio ports.
- Use same device selection flow for input as for outputs:
getNewInputDevice -> getDeviceForInptusiource -> setInputDevice
Change-Id: Idaa5a883b19a45816651c58cac697640dc717cd9
ndroid.mk
udioFlinger.cpp
udioFlinger.h
atchPanel.cpp
atchPanel.h
hreads.cpp
hreads.h
|
| 1a83b736737e7f625371093519ff7a10b05e0c63 |
27-May-2014 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: fix deadlock upon AudioRecord creation error"
|
| 4b123406c10c17852734a1b691bb9ce2a4cb7caf |
11-Apr-2014 |
Eric Laurent <elaurent@google.com> |
IAudioFlinger interface extension for patch panel
Change-Id: Iaabe0a7e315d5725e00a74a6ed40339b98f20330
udioFlinger.h
|
| 5912117110ae6efadd862fcb0e395fdcd18aa00c |
27-May-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Move validation of frameCount from set to openRecord_l"
|
| fe1a94e68e173fe4dfe7699112422a94eddacb4e |
27-May-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix deadlock upon AudioRecord creation error
AudioFlinger:openRecord() should not hold mClientLock when
releasing the local reference on AudioRecord as the destructor will
also lock mClientLock.
Same fix for AudioFlinger::createTrack().
Also make sure that AudioFlinger::createEffect() holds mClientLock
when clearing local reference to the Client in case of error.
Regression introduced by 021cf9634ab09c0753a40b7c9ef4ba603be5c3da
Bug: 15118096.
Change-Id: Ie961c398c8e0460bca9b95e2ee4ce6859316c275
udioFlinger.cpp
|
| 45d68d366d07b10733c1e8af6a6984befbc08ce1 |
24-May-2014 |
Andy Hung <hunga@google.com> |
Rename mixBuffer to mMixerBuffer in FastMixer
Likewise mixBufferState becomes mMixerBufferState.
This harmonizes with the naming in AF::MixerThread.
Change-Id: I1255d7c07cc2c6ee925d7430925236d2bd163122
astMixer.cpp
astMixer.h
|
| ef7c7fbd0e3fb36af14cd7d39f64c949031516a5 |
13-May-2014 |
Andy Hung <hunga@google.com> |
Add multiple format capability to AudioMixer
Change-Id: I04ac1cafd90b6ed652f8d51888ad07576678f0bc
Signed-off-by: Andy Hung <hunga@google.com>
udioMixer.cpp
udioMixer.h
astMixer.cpp
|
| c62476f0c0c1cf9283a38852bde0a4c9434df712 |
24-May-2014 |
Andy Hung <hunga@google.com> |
Merge "Add format parameter to getTrackName() and track_t"
|
| 39cf5681975ec9097742c47de7c2d0f8c11248a2 |
23-May-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Allow track buffer "allocation" to be from pipe"
|
| 24ffe7a9161943c8ba2353e956653a71af564b5d |
23-May-2014 |
Elliott Hughes <enh@google.com> |
am 79262cfc: am 5b756bfd: am 8b187f4a: Merge "Move frameworks/av off <sys/atomics.h>."
* commit '79262cfc14a9d8b5aa29d43e84ea91eedb9492ba':
Move frameworks/av off <sys/atomics.h>.
|
| f91df1b368a140abd37c80b204bd48d78778cc43 |
13-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Start adding FastCapture based on FastThread WIP
This version supports at most one fast capture client.
Change-Id: Idf609bfc80ae22433433d66a5232c043c65506df
ndroid.mk
astCapture.cpp
astCapture.h
astCaptureState.cpp
astCaptureState.h
tateQueueInstantiations.cpp
|
| 29b703eec27b305e7b5b2343bf257643e38f6b68 |
12-May-2014 |
Glenn Kasten <gkasten@google.com> |
Move validation of frameCount from set to openRecord_l
This move is needed because frameCount is validated on server side for
fast tracks (as should be done for normal tracks too).
Change-Id: I6d99e80869fd90fab373cf60ef348c01f075fbca
hreads.cpp
|
| 6181ffd90a436f333c43a7f812391eee2c35005a |
13-May-2014 |
Glenn Kasten <gkasten@google.com> |
Allow track buffer "allocation" to be from pipe
Change-Id: Ib9ac170f8e8b7746b3588157a56cbee3b753a1cb
hreads.h
rackBase.h
racks.cpp
|
| 5b756bfdaa27c8b606720544bf507213cf47376d |
23-May-2014 |
Elliott Hughes <enh@google.com> |
am 8b187f4a: Merge "Move frameworks/av off <sys/atomics.h>."
* commit '8b187f4acea8d7e517ff9ced32fc7368bdbaa6e0':
Move frameworks/av off <sys/atomics.h>.
|
| 63492411cf4abff452b47f59ec7ff6f3e9925c84 |
22-May-2014 |
Elliott Hughes <enh@google.com> |
Move frameworks/av off <sys/atomics.h>.
Bug: 14903517
Change-Id: I92cdfc618d83fd914cfb558a07b1deae0255fccb
astMixer.cpp
|
| 5806b3533437e3d83208e8e9d6bd74ed304e51ec |
22-May-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: test min HAL version instead of current
Change-Id: Ibce2353b4390329b4c1a453e829178e3986c25a7
udioFlinger.cpp
|
| 87cebadd48710e42474756fc3513df678de045ce |
22-May-2014 |
Elliott Hughes <enh@google.com> |
resolved conflicts for merge of f285c91e to master
Change-Id: I4c995e297d09c0854a42142cc57dce8b771acf65
|
| c10b874c4d0d6208fb93a7aa0dc8af4418faad45 |
22-May-2014 |
Calin Juravle <calin@google.com> |
am d5516339: Merge "Move frameworks/av from fdprintf to POSIX dprintf."
* commit 'd551633916d3677da4dda10aba6837bdc6803ee0':
Move frameworks/av from fdprintf to POSIX dprintf.
|
| 8b5f642eb2364ea7fe46a5b3af51b48b58f12183 |
22-May-2014 |
Elliott Hughes <enh@google.com> |
Move frameworks/av from fdprintf to POSIX dprintf.
Bug: 11156955
Change-Id: Ia98cd16b4c1f7be87cf060b7456de4f40896bacb
udioFlinger.cpp
udioWatchdog.cpp
astMixer.cpp
tateQueue.cpp
hreads.cpp
|
| 9a59d411cbe68d6027c541854c6d7c1348e4ee3d |
22-May-2014 |
Elliott Hughes <enh@google.com> |
am 85e6e87c: Merge "Restore accidentally-removed NULL timeout."
* commit '85e6e87c41bba51729cee8958a5c30157c8dc674':
Restore accidentally-removed NULL timeout.
|
| e348c5b72ad889389c7c1c900c121f0fbee221b5 |
22-May-2014 |
Elliott Hughes <enh@google.com> |
resolved conflicts for merge of c0bf836f to master
Change-Id: I5cf238fa56d6fff443b212870cf9dd7f1110e7da
|
| 22f22ffcb111b38eb7a303bfde39aed108b43e07 |
22-May-2014 |
Elliott Hughes <enh@google.com> |
Restore accidentally-removed NULL timeout.
Change-Id: I933c80fea50950243f109acf8d6350013aaa811f
astMixer.cpp
|
| ad71caf25e31ecfc4ecf2c925e38fd87905fbc8e |
22-May-2014 |
Elliott Hughes <enh@google.com> |
am 3b6da6bf: Merge "Move frameworks/av off private API."
* commit '3b6da6bfb50a1c0324fb4c87326d3a477944d5fe':
Move frameworks/av off private API.
|
| ee499291404a192b059f2e04c5afc65aa6cdd74c |
22-May-2014 |
Elliott Hughes <enh@google.com> |
Move frameworks/av off private API.
Bug: 11156955
Change-Id: Ib3bb9d66a2bf92977c2445b62aa074a2d7a45aea
astMixer.cpp
astMixer.h
hreads.cpp
racks.cpp
|
| 9382378b6b271949bdce527ea3da748c13c4c692 |
22-May-2014 |
Elliott Hughes <enh@google.com> |
resolved conflicts for merge of b1080917 to master
Change-Id: Iabbe3e95e29f28461ff722b550e8a9f35bd2528f
|
| acbd9492996348bf4565fbaa5a14772f0f4ace38 |
22-May-2014 |
Elliott Hughes <enh@google.com> |
am d084206d: Merge "Move frameworks/av off __futex_syscall3."
* commit 'd084206d5a4a5364211af46e4201311933e83bfc':
Move frameworks/av off __futex_syscall3.
|
| 77e1512aee8092069514973379a59a318e721be3 |
22-May-2014 |
Elliott Hughes <enh@google.com> |
Move frameworks/av off __futex_syscall3.
Bug: 11156955
Change-Id: I943080bc51b6efa781059b48b5b5ecc4fa287512
hreads.cpp
racks.cpp
|
| 00ddac6f47a066a545342a81c40425e3a153e8a1 |
20-May-2014 |
Andy Hung <hunga@google.com> |
Merge "Update AudioMixer::getTrackName to save channelMask"
|
| b4db73d022f3de3530bc2b3c9c831ccfdd1a2ead |
20-May-2014 |
Andy Hung <hunga@google.com> |
Merge "Use new channel count functions for audio masks"
|
| 1e54e6dfc1bb8d33089ed8857f0f194e696f85e2 |
19-May-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Pass stereo gains as packed minifloat"
|
| e8a1ced4da17dc6c07803dc2af8060f62a8389c1 |
10-May-2014 |
Andy Hung <hunga@google.com> |
Add format parameter to getTrackName() and track_t
Change-Id: Ia152a839014e235fbfb656104c15d7c1b456d02e
Signed-off-by: Andy Hung <hunga@google.com>
udioMixer.cpp
udioMixer.h
astMixer.cpp
astMixerState.cpp
astMixerState.h
hreads.cpp
hreads.h
racks.cpp
|
| c56f3426099a3cf2d07ccff8886050c7fbce140f |
22-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Pass stereo gains as packed minifloat
This will allow (eventually) a greater dynamic range for gains.
However there are still a few remaining places in effects and mixer
that will also need to be changed in order to get the full benefit.
Also fixes a minor bug: was not checking for NaN in AudioTrack C++.
Change-Id: I63bce9e82e0a61546d8ff475fb94bcb700d99c96
udioFlinger.h
udioMixer.h
ffects.h
astMixer.cpp
astMixerState.h
laybackTracks.h
hreads.cpp
racks.cpp
|
| 5859b98476de3bbb4ae4dc00d4b2f010ca7897d3 |
19-May-2014 |
Eric Laurent <elaurent@google.com> |
am be2664da: am 4c9bde7a: am ce473c32: Merge "Codec Offload: Add NULL check as mCallbackThread is NULL when offloaded HAL is running BLOCKING mode"
* commit 'be2664dac34a6a9d9f754d4097b32158d4947cd2':
Codec Offload: Add NULL check as mCallbackThread is NULL when offloaded HAL is running BLOCKING mode
|
| 4c9bde7a3e636d3ecccd7905d142dbbb80e977a0 |
19-May-2014 |
Eric Laurent <elaurent@google.com> |
am ce473c32: Merge "Codec Offload: Add NULL check as mCallbackThread is NULL when offloaded HAL is running BLOCKING mode"
* commit 'ce473c32705e2d1c5fcc817198101e1220b3c8f7':
Codec Offload: Add NULL check as mCallbackThread is NULL when offloaded HAL is running BLOCKING mode
|
| 68112fc4f77ab8c4a744782f78e9792afe0cbfc1 |
14-May-2014 |
Andy Hung <hunga@google.com> |
Update AudioMixer::getTrackName to save channelMask
Change-Id: I4fb4abbff1b05bdd0f6e21b5f5f4c3f2a57ae28d
Signed-off-by: Andy Hung <hunga@google.com>
udioMixer.cpp
astMixer.cpp
|
| e541269be94f3a1072932d51537905b120ef4733 |
16-May-2014 |
Andy Hung <hunga@google.com> |
Use new channel count functions for audio masks
Change-Id: Ia658ab4b6320d19fdb50f123c930918724ff0ef3
Signed-off-by: Andy Hung <hunga@google.com>
udioFlinger.cpp
udioMixer.cpp
hreads.cpp
racks.cpp
|
| deb9081a61b446dbe50461082974a4228899f178 |
14-May-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Pull up mFlags and isFastTrack from Track to TrackBase"
|
| cbe6fddebe3ec84176037de7f9681d2407fa1113 |
14-May-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Explicitly include Configuration.h"
|
| 2d859b91a59b4f6d63ff2bd93b462cab60f4a8d7 |
14-May-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix memory corruption if increaseSamplingN is not called"
|
| 56604aa3a56dc8e15532597a0a74b3c7b165e006 |
13-May-2014 |
Uday Gupta <guptau@broadcom.com> |
Codec Offload: Add NULL check as mCallbackThread is NULL
when offloaded HAL is running BLOCKING mode
b/14897952
Change-Id: I159a33e1181c1a62d80ab4bd3337a35dafaad884
Signed-off-by: Uday Gupta <guptau@broadcom.com>
Signed-off-by: Pierre Couillaud <pierre@broadcom.com>
hreads.cpp
|
| 755b0a611f539dfa49e88aac592a938427c7e1b8 |
13-May-2014 |
Glenn Kasten <gkasten@google.com> |
Pull up mFlags and isFastTrack from Track to TrackBase
Change-Id: I6bd48292310be4e05758a11f617f50585904422d
laybackTracks.h
ecordTracks.h
hreads.cpp
rackBase.h
racks.cpp
|
| 5b17c0b5b418c340d3e5201a72ee8c88c1755355 |
13-May-2014 |
Glenn Kasten <gkasten@google.com> |
Explicitly include Configuration.h
Must include Configuration.h in any source file with #ifdef or #if that
depends on Configuration.h. This avoid inconsistencies that could result
in subtle bugs.
Change-Id: I99fdf19d56e7c73de4e7d672b937336b932a2a00
udioFlinger.h
tateQueue.h
|
| 963698408b1ce0f2ac7eab1e863dc7271de116a3 |
13-May-2014 |
Glenn Kasten <gkasten@google.com> |
Fix memory corruption if increaseSamplingN is not called
Change-Id: Icbf9aaf68c25ba3330ceb79167bd776cf0342d0d
astThreadState.cpp
|
| 021cf9634ab09c0753a40b7c9ef4ba603be5c3da |
13-May-2014 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: add specific mutex for client lists
Add a specific mutex to protect access to mClients and
mNotificationClients lists. This avoids locking the main AudioFlinger
mutex from inside thread loops and allows not to worry about
cross deadlocks when sending a config event with status reply while
keeping the ThreadBase or AudioFlinger mutex locked.
As a way of consequence, remove notification client list passed to
processConfigEvents_l() and audioConfigChanged() as the list
can now be accessed by locking client mutex only.
Change-Id: I228022204b6709a8bb60cc96d9514a6ffe59b62e
udioFlinger.cpp
udioFlinger.h
ffects.cpp
hreads.cpp
hreads.h
racks.cpp
|
| 110135b9610fee8bfa5bac2d9ef4fc4c24991c36 |
09-May-2014 |
Glenn Kasten <gkasten@google.com> |
Explicitly ignore return value of __futex_syscall3
Change-Id: Ic65f257bfc4a0eff7bdd3c44fab68a3f17e2bd08
hreads.cpp
|
| 0fdbdd2a3909e4692fa3baaaa1f53eb91b31af56 |
10-May-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove obsolete IAudioRecord::getCblk()"
|
| 26d5ff926fa3323b39ae4408bcd29826a9523c9b |
10-May-2014 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: refactor thread config events"
|
| fe9570c7b937b49d3603ccb394aed732b79bc6be |
07-May-2014 |
Glenn Kasten <gkasten@google.com> |
Remove obsolete IAudioRecord::getCblk()
Change-Id: Id20b5efd765b9796b0e391610e06dc928a829ebf
udioFlinger.h
racks.cpp
|
| 1035194cee4fbd57e35ea15c56e66cd09b63d56e |
09-May-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: refactor thread config events
Merge config events and set parameters messaging mechanism.
- setting parameters now uses the config event mechanism
- config event now allow to wait for a condition and synchronize caller
binder thread with execution thread and return an execution status.
- simplify locking mechanism to avoid unlocking the thread
mutex while processing events.
Change-Id: Ia49cb3e617abec4bacb6c1f9a8cb304c4ed3902e
udioFlinger.cpp
udioFlinger.h
hreads.cpp
hreads.h
|
| d776ac63ce9c013c9626226e43f7db606e035838 |
07-May-2014 |
Glenn Kasten <gkasten@google.com> |
IAudioFlinger::openRecord returns IMemory(s)
openRecord() now explicitly returns the control block and data buffer
as separate IMemory references. If the IMemory for data buffer
is 0, this means it immediately follows the control block.
Change-Id: Ic098f88f0e037f8fbe30006689e18cacacf09d06
udioFlinger.cpp
udioFlinger.h
ecordTracks.h
hreads.cpp
rackBase.h
racks.cpp
|
| b880f5e5fc07397ddd09a94ba18bdf4fa62aae00 |
07-May-2014 |
Glenn Kasten <gkasten@google.com> |
Add a per-thread heap that is read-only to clients
Change-Id: I401263566ca20fbfb565689c8fa99458d3b283b2
hreads.cpp
hreads.h
|
| 828f883a43f66f77d776a75d0ea2b87c7c826071 |
07-May-2014 |
Glenn Kasten <gkasten@google.com> |
Fix a couple of fast capture typo bugs
AudioRecord constructor was not passing flags through to set().
Server-side check was using wrong kind of channel mask.
Change-Id: Ifaa880ec323771e9fd168262be05f3e539f53390
hreads.cpp
|
| f655acf15e587dbf5d7120e9f2bf4b5bcb89941c |
06-May-2014 |
Eric Laurent <elaurent@google.com> |
am 8fb04d47: Merge "Fix memory leak when filtering commands in insertCommand_l()"
* commit '8fb04d474ed9e0d1302ae223ee76fafbe5fb41a3':
Fix memory leak when filtering commands in insertCommand_l()
|
| 8fb04d474ed9e0d1302ae223ee76fafbe5fb41a3 |
06-May-2014 |
Eric Laurent <elaurent@google.com> |
Merge "Fix memory leak when filtering commands in insertCommand_l()"
|
| d8484dd4cf124854e7ff950da4300887f32fd05b |
05-May-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Use audio_is_linear_pcm()"
|
| c4b88a8d0f524666bf0f390075c334d047a104f2 |
01-May-2014 |
Glenn Kasten <gkasten@google.com> |
Fix bug for direct track with PCM != 16-bit
The AUDIO_FORMAT_PCM_8_BIT format was being converted to
AUDIO_FORMAT_PCM_16_BIT on client side even for direct tracks.
That conversion was incorrect; it should only be done for mixed tracks.
Also remove checks for specific PCM formats in the generic part of
server side of createTrack. Those format checks should only be done by
the thread. This will allow direct tracks for PCM 8-bit, PCM 24-bit, etc.
Change-Id: If5b9fd79f8642ed93e2aeabcaf4809b2ed798978
udioFlinger.cpp
|
| 993fa0603707e94ce259e95e56838a85b5ccbdc5 |
02-May-2014 |
Glenn Kasten <gkasten@google.com> |
Use audio_is_linear_pcm()
Change-Id: I17b30fdb6dbb6454ad0a20dd703fd603a37a8397
hreads.cpp
|
| 1a7be1ec9c769203e6c8e26378de0ab8e2ad493d |
29-Apr-2014 |
Bill Yi <byi@google.com> |
Merge commit '91820d46b5f3065c2fded3cdf65d305715b33bb1' into HEAD
|
| 2234002b0710c8db73f82d397cb945cd541c6bbb |
07-Apr-2014 |
Glenn Kasten <gkasten@google.com> |
Start pulling bits of FastMixer up to FastThread
Change-Id: I4c6f7b8f88fcf107bb29ee6432feecd4ab6554d2
ndroid.mk
onfiguration.h
astMixer.cpp
astMixer.h
astMixerDumpState.h
astMixerState.cpp
astMixerState.h
astThread.cpp
astThread.h
astThreadState.cpp
astThreadState.h
|
| e273141263e988ab086a8d41e0e4fc75c00f1b86 |
24-Apr-2014 |
Glenn Kasten <gkasten@android.com> |
am 7e5c6351: am 4513aa2c: Merge "AArch64: rewrite audioflinger\'s sinc resample by intrinsics."
* commit '7e5c635114a7762ad44581300021667a0da97389':
AArch64: rewrite audioflinger's sinc resample by intrinsics.
|
| 7e5c635114a7762ad44581300021667a0da97389 |
24-Apr-2014 |
Glenn Kasten <gkasten@android.com> |
am 4513aa2c: Merge "AArch64: rewrite audioflinger\'s sinc resample by intrinsics."
* commit '4513aa2cda9e636e4ac675dab9a1353b22e951ae':
AArch64: rewrite audioflinger's sinc resample by intrinsics.
|
| 4513aa2cda9e636e4ac675dab9a1353b22e951ae |
24-Apr-2014 |
Glenn Kasten <gkasten@android.com> |
Merge "AArch64: rewrite audioflinger's sinc resample by intrinsics."
|
| 1b24cf32e02a1372e57b3a2a432bc0adfe496e6d |
16-Apr-2014 |
Mark Salyzyn <salyzyn@google.com> |
am e80631aa: am 839d11d1: Merge changes I0a744dc7,Id993a70d
* commit 'e80631aa1992ca50af679cd6a018c0ffda7f9b17':
media: use size_t for integer iterator to Vector::size()
media: 64 bit compile issues
|
| e80631aa1992ca50af679cd6a018c0ffda7f9b17 |
16-Apr-2014 |
Mark Salyzyn <salyzyn@google.com> |
am 839d11d1: Merge changes I0a744dc7,Id993a70d
* commit '839d11d1f7be9dff2f06c7d30a9eb39cb6782078':
media: use size_t for integer iterator to Vector::size()
media: 64 bit compile issues
|
| 3ab368e0810d894dcbc0971350c095049478a055 |
15-Apr-2014 |
Mark Salyzyn <salyzyn@google.com> |
media: use size_t for integer iterator to Vector::size()
Change-Id: I0a744dc7815a86a993df9b0623440be620ec8903
udioFlinger.cpp
racks.cpp
|
| 96b0d5ef85d4d71690c41c84456afbf19864c785 |
15-Apr-2014 |
Andy Hung <hunga@google.com> |
Merge "Improve resampler filter computation speed"
|
| c125f38cd0ae35409a01b98a99e483550daa1313 |
12-Apr-2014 |
Glenn Kasten <gkasten@google.com> |
audioflinger: forward device change to effects in direct thread
Device change during offload playback is not informed to effect chain
which causes effects still work on unexpected output devices. Add device
change notification in direct output thread.
Original author wjiang <wjiang@codeaurora.org>
CRs-Fixed: 630408
Bug: 14053172
Signed-off-by: Glenn Kasten <gkasten@google.com>
Change-Id: I094a99bdf540479cee2fca6614ec35c2fa7d6046
hreads.cpp
|
| 48412dc92c4cc92a3f4821616e12290f5eeabd07 |
24-Mar-2014 |
Jesper Tragardh <jesper.tragardh@sonymobile.com> |
Fix memory leak when filtering commands in insertCommand_l()
When AudioPolicyService::AudioCommandThread::insertCommand_l filters
a command from the queue, the AudioCommand* is removed from the
vector mAudioCommands by calling mAudioCommands.removeAt(). This will
destroy the pointer and compact the vector, but not delete the
AudioCommand object.
This patch adds code to delete a filtered out AudioCommand and its
data object mParam. However, mParam was void* and deleting a void
pointer is undefined behavior according to the C++
standard. Therefore, the data classes are modified to inherit
from an abstract base class with a virtual destructor.
The deletion of non-filtered commands in
AudioPolicyService::AudioCommandThread::threadLoop() was already
correct, as each case casts mParam to the proper type before
deleting.
This also fixes AOSP issue 36370.
Change-Id: I2b63838aff3608132aa496f964a929f78c4b6267
udioPolicyService.cpp
udioPolicyService.h
|
| df383a5a54582811e5e038efc557172b8ec69dd1 |
10-Apr-2014 |
Andy Hung <hunga@google.com> |
Update test-resample to handle multichannel
Option -c # specifies number of channels (mono default).
Option -s to specify stereo is removed (-c 2 replaces).
Option -h to specify WAV header is removed (WAV is now default).
Change-Id: Iba4b83806028a8a9c1ddba6f555182d214ef73ff
Signed-off-by: Andy Hung <hunga@google.com>
est-resample.cpp
|
| 12b44bd5fe3069cd3450d05b6c446b600e0553d3 |
10-Apr-2014 |
Zhongwei Yao <zhongwei.yao@arm.com> |
AArch64: rewrite audioflinger's sinc resample by intrinsics.
Pass conformance test on armv7 and aarch64, performance test is done on armv7.
Compared with original armv7 assembly version, this version has similar
result. Here is performance data on pandaboard android4.4 (input: random wave
file, unit: Mspl/s, toolchain:gcc 4.8):
| | origin(assembly) | current(intrinsics) | C version |
|----------------+------------------+---------------------+-----------|
| single channel | 6.17 | 7.14 | 3.43 |
| double channel | 5.24 | 5.63 | 3.50 |
Change-Id: If5670218e1586e9dfd2b8d9c66a6880f3e4808ca
udioResamplerSinc.cpp
|
| 771386e6e6e79697e2d839ef0f25a242946ba1e5 |
09-Apr-2014 |
Andy Hung <hunga@google.com> |
Add and enable floating point option for audio resampler
Can be tested with test-resample.
Change-Id: I8339846d7c647444b6025d33cfa145d5d3658121
Signed-off-by: Andy Hung <hunga@google.com>
udioResampler.cpp
udioResamplerDyn.cpp
udioResamplerDyn.h
|
| d549139155b20d7cbf6a4326133e06def465ef54 |
09-Apr-2014 |
Andy Hung <hunga@google.com> |
Add floating point to audio resample processing
Add template type parameters for input, output data type.
Minor change in non-NEON mono channel handling.
Minor fixup on comments.
Change-Id: I7dc9972d130913718b62f32c02d31f99c06682f2
Signed-off-by: Andy Hung <hunga@google.com>
udioResamplerDyn.cpp
udioResamplerFirProcess.h
|
| 430b61c72094882bc48693dfc10c256a6ae36ee9 |
09-Apr-2014 |
Andy Hung <hunga@google.com> |
Add capability for floating point Kaiser window filters
Change-Id: Idbb33248bbab2300c2650a4657d8fbc482a5d46c
Signed-off-by: Andy Hung <hunga@google.com>
udioResamplerFirGen.h
|
| 781366833a12877b8d5ad4aa081114e30f799319 |
09-Apr-2014 |
Andy Hung <hunga@google.com> |
Update test-resample to handle floating point
Change-Id: Ib34d716fbabcd5eb70f8a5ffcf362e242671d916
Signed-off-by: Andy Hung <hunga@google.com>
est-resample.cpp
|
| 0d521d91154fe0199181c845b6dc70faf38ca8fb |
08-Apr-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix 64-bit build warnings"
|
| 1e4e4f40b4e2c655ca377adbba53df897545c3ae |
08-Apr-2014 |
Glenn Kasten <gkasten@google.com> |
Fix 64-bit build warnings
Change-Id: I667401522cb4ccd41013e2883a4c75ddeca08ef6
est-resample.cpp
|
| bc6ae9bd4ee93e91e212ad35be038f21797ef31e |
07-Apr-2014 |
Glenn Kasten <gkasten@google.com> |
Fix regression in FastMixer statistics
At some point, the mSampleRate in FastMixerDumpState stopped being
initialized correctly. I'm not sure when this happened, it doesn't seem to
be introduced in recent CLs. This lack of initialization caused
some of the FastMixer statistics based on sample rate to be useless.
Change-Id: Id2a96d606130a90c4c4f1bddd59778f6c6428a9c
astMixer.cpp
|
| 84a0c6e87c48f58a0d3be71961432c086a4d24cc |
02-Apr-2014 |
Andy Hung <hunga@google.com> |
Change references of Q19.12 to Q4.27 for clarity
Change-Id: I5beb7daf6ff9bc123ff3582f7c294edcaf8652f6
Signed-off-by: Andy Hung <hunga@google.com>
udioMixer.cpp
udioResampler.h
ecordTracks.h
hreads.cpp
est-resample.cpp
|
| bafa561d0c9363c5307b6b1daa498bd3ae36089d |
02-Apr-2014 |
Andy Hung <hunga@google.com> |
Improve resampler filter computation speed
Approximate speed improvement is 10-15% for filter computation,
which is floating point intensive. This will be important
for devices without hw floating point support.
Change-Id: I10b4e778c8d632b52218a777504b092c189e437f
Signed-off-by: Andy Hung <hunga@google.com>
udioResamplerFirGen.h
|
| 2ade30003882f0c3782d68018d06cb81b99dfeb2 |
01-Apr-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "FastMixer use FastThread and FastMixerState use FastThreadState"
|
| 1e0c8cd8fad2108c8ea085748fcca2059e3ce4ab |
01-Apr-2014 |
Glenn Kasten <gkasten@google.com> |
Merge changes I5748f47d,I3f074933
* changes:
Add FastThread.h
Add FastThreadState
|
| f7160b5dc3091fd9624f25d534c6848ee831b706 |
19-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
FastMixer use FastThread and FastMixerState use FastThreadState
Change-Id: I594c973e9f575113bdefee6f4cf8c29d8beac1f3
astMixer.h
astMixerState.cpp
astMixerState.h
|
| 4cd161b5bf7a8008d684db4b217ed84d1704e683 |
19-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Add FastThread.h
Change-Id: I5748f47dbfa42c14cc93973742e05ac963bf3ba8
astThread.h
|
| a18f644ce5eb4ab876b343e24ea613566c00058f |
19-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Add FastThreadState
Change-Id: I3f07493375ace6e5cfdcd02ad90c4b6fad543b0c
ndroid.mk
astThreadState.cpp
astThreadState.h
|
| 3a6c90aa0617666d9abc94c02b752d9eb3d64772 |
13-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Add FIXMEs to createRecord for fast capture
Change-Id: I759be200fae32969212c52a409f46f2e704081e3
hreads.cpp
hreads.h
|
| 1490d5142d8d485ea0d56a9974726fb6dff0a62d |
28-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Mark the logging heap read-only to media.log service"
|
| c9b8d90dabdf1234be0117c592f635a92c4eb0d9 |
27-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Use symbol AUDIO_IO_HANDLE_NONE from <system/audio.h>"
|
| 5c4fa907328c8ee586367272c7525cb87742f14d |
26-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Use LOG_ALWAYS_FATAL instead of LOG_FATAL"
|
| 142f519aa1acd5804d111e60d100f170fed28405 |
26-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Use symbol AUDIO_IO_HANDLE_NONE from <system/audio.h>
Change-Id: Id6b1aa17558eb73e17f22b8eab6cd02e00a96dff
udioFlinger.cpp
|
| adad3d7d935da176ff24941b4ae9edf7340e9b96 |
21-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Use LOG_ALWAYS_FATAL instead of LOG_FATAL
LOG_FATAL is compiled out in most builds, so the
assertion checks were not being performed.
Change-Id: I774f0985ab9c5ccecd8989a0f1c940386b73fc35
udioMixer.cpp
astMixer.cpp
hreads.cpp
racks.cpp
|
| ecdca392ed40e89a6ef14b86446e952c35497ac4 |
26-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Use symbolic constants from <system/audio.h>"
|
| 51beebba9d4989d05831a13bbe5cdf53ecac1bf5 |
26-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Use symbol AUDIO_DEVICE_NONE from <system/audio.h>"
|
| 73ec11738cd4399c55d4905665a14a7c4cd42cbf |
26-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix uses of KeyedVector"
|
| 45faf7e02791993a487d6e038d16ff46395f1975 |
17-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Use symbol AUDIO_DEVICE_NONE from <system/audio.h>
Change-Id: I61f882c5e7c949bf00d3bfc745ebf3b5e1c42a58
udioFlinger.cpp
|
| 66e4635cb09fadcaccf912f37c387396c428378a |
17-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Use symbolic constants from <system/audio.h>
AUDIO_INTERLEAVE_*
AUDIO_STREAM_MIN
AUDIO_SESSION_ALLOCATE
Change-Id: I31dd6f327204685e50716079ce21c4ba206dff11
hreads.cpp
|
| 9ea65d0f4a564478343b1a722fae4ce5883670c3 |
17-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Fix uses of KeyedVector
Constructor for AudioFlinger::mAudioHwDevs was missing, and so
AudioFlinger::findSuitableHwDev_l() could return an undefined pointer
if a non-0 module wasn't found.
A KeyedVector of Plain Old Data (POD) element type must specify the
default value in the constructor, or else the default will be undefined.
Minor:
- Parameter had wrong type in constructor for AudioSystem::gOutputs.
- Remove obsolete AudioSystem::gStreamOutputMap.
Change-Id: I9841493e018440e559d8b8b0e4e748ba2b2d365b
udioFlinger.cpp
|
| bcefec31bd1346133052356ffc8d7ac8a5b13fab |
17-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Document AudioFlinger::nextUniqueId()
Change-Id: Iafe96f1c10bd85cb23a2553945ca68aa601dc2eb
udioFlinger.cpp
udioFlinger.h
|
| 46d1b4ab7e83c677532587c7c8bd46eb8b30fc67 |
19-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
resolved conflicts for merge of f40c4c56 to master
Change-Id: Ifd5385ad42a81e02e6a6afc6281f09fbff361671
|
| d09ec3b22013ef43d422c916dfd44bab9101a659 |
19-Mar-2014 |
Glenn Kasten <gkasten@android.com> |
am 36817364: Merge "Add libaudioresampler"
* commit '36817364738d4c45adc3e448fbec02a9611bfeda':
Add libaudioresampler
|
| f40c4c561151ae93595e8fab5dcaaa1c69df74ef |
19-Mar-2014 |
Glenn Kasten <gkasten@android.com> |
am 36817364: Merge "Add libaudioresampler"
* commit '36817364738d4c45adc3e448fbec02a9611bfeda':
Add libaudioresampler
|
| ddf887c08b58382920effb7f0b2049e471960b4a |
19-Mar-2014 |
Glenn Kasten <gkasten@android.com> |
Add libaudioresampler
libaudioresampler is available in both 32-bit and 64-bit,
unlike libaudioflinger which is currently 32-bit only.
Bug: 8141282
Change-Id: I839f7b4e6aaed6984012ca6d514323f927669df6
ndroid.mk
|
| cf515ab0030cf5e0bef843638af7c989a8f6843a |
15-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Mark the logging heap read-only to media.log service
Change-Id: Ie26a9e7e37c951774c71d2c53886db52dd5479aa
udioFlinger.cpp
|
| d8590167fec151cc2e8f6e28e406c99882f40884 |
18-Mar-2014 |
Glenn Kasten <gkasten@android.com> |
am 77658a06: Merge "AudioPolicyService: malloc/delete pair"
* commit '77658a069c81a0d5b4a1b81443b470a3ea64cab0':
AudioPolicyService: malloc/delete pair
|
| 0c282de73240b1efce64558ab2fa0a8be1f4292b |
18-Mar-2014 |
Glenn Kasten <gkasten@android.com> |
am 77658a06: Merge "AudioPolicyService: malloc/delete pair"
* commit '77658a069c81a0d5b4a1b81443b470a3ea64cab0':
AudioPolicyService: malloc/delete pair
|
| 77658a069c81a0d5b4a1b81443b470a3ea64cab0 |
18-Mar-2014 |
Glenn Kasten <gkasten@android.com> |
Merge "AudioPolicyService: malloc/delete pair"
|
| 2d3ca68363f723fbe269d3ce52dab4985dfc7154 |
07-Mar-2014 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audioflinger: OffloadThread fix for resume underrun
AudioFlinger can miss resuming h/w on a pause->resume transition
if sufficient data isn't available
Bug: 11358524.
Change-Id: Ic3c75256290d3515fd4a96dfcc900909fbe5bc15
hreads.cpp
racks.cpp
|
| 164985121796cf214c7a83d32005d9b01125b558 |
18-Mar-2014 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: Change track ready condition in STOPPING_1 state
Set FS_FILLED only if framesReady() > 0
Change-Id: Ibb1d2e988ff17fcf3a7ab61031a3f85df82f18d5
racks.cpp
|
| 8d6c292a0bed3d63b5b7297d09a604af6327c663 |
05-Feb-2014 |
Krishnankutty Kolathappilly <kkolat@codeaurora.org> |
audioflinger: Handle pause /resume in stopping state
-Pre-requisite:
Perform seek on the clip. After seek the data remaining till EOS
is little more than the driver and common block buffering.
-Framework state:
Offload thread is waiting for signal from the HAL for a free
buffer. Audio Player calls sink stop on reaching EOS. Audio
track is waiting on obtain buffer for a free space in common
block to send the last buffer. The track is moved to stopping
state as input EOS is reached.
-Issue:
Perform pause/ resume in this state(STOPPING), Audio playback
does not resume.
-Fix
Ensure resume is called in stopping state if frames ready is
greater than zero.
Bug: 12870871
Change-Id: Ib1378c4ee5ce4bea655691e93de0775f7b1d2804
Signed-off-by: Glenn Kasten <gkasten@google.com>
laybackTracks.h
hreads.cpp
racks.cpp
|
| 314869117d5e3365fc3e43fee01209d18515ca42 |
18-Mar-2014 |
Eric Laurent <elaurent@google.com> |
Merge "move audio policy service to a separate library"
|
| 3ef14ef30359376006a233f6a21a165d4b65a7df |
13-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Make error handling more similar for output and capture
And simplify error case
Change-Id: I0bb1ec252945d672cc4cef137977b912f1b23d51
udioFlinger.cpp
racks.cpp
|
| 15e5798908ccac14e10c84834eaf08c42931bd06 |
24-Sep-2013 |
Glenn Kasten <gkasten@google.com> |
Make error handling for audio driver initCheck more similar
Change-Id: I8222f00f7ac89d52235af0f64f3fee7a8c471e1e
hreads.cpp
|
| 570f633e0b02d1bc25f3312b92e72cc29a40ca38 |
13-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Make session ID code similar between output and capture
Change-Id: I873a97d0e36e1a452421df4f749d51e1916797a5
udioFlinger.cpp
|
| e848bd9abb3bbbd4c66f9fb9d1442f7663d7ba40 |
13-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Move declarations to where needed
Also remove unnecessary ALOGV and dead variable inFrameCount
Change-Id: I34547fb0fd2ff142fc60277bedfa4064e7356611
udioFlinger.cpp
|
| c5a17425986b4ce3384e6956762c86018b49c4a0 |
13-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Remove name output parameter from createTrack
It was only used for one log.
A better solution will be a per-track unique ID.
Change-Id: Ia440e02ae4a5a4019a9a2d08970e1ee93ac4c3a3
udioFlinger.cpp
udioFlinger.h
|
| c3df838434b37d8400eea2438083cc01a4c1cc71 |
13-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Replace if/else by switch
Change-Id: Ibe96c8fb1f0ba7f850c0561f60769ce63aceba5d
hreads.cpp
|
| 05997e21af6c4517f375def6563af4b9ebe95f39 |
13-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Whitespace and comments
Change-Id: I99752cca74a459c4746ea4718acb0115439e50bd
udioFlinger.cpp
hreads.cpp
racks.cpp
|
| 0e0e846f3d86906ff990025981e514d6bcf44d8e |
13-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Prepare for additional capture formats
Change-Id: Ib10b004f0029191b33a97671e41b0d86ebfbe0b9
udioFlinger.cpp
|
| 758799cef41fc5495f59eeb6249d02ce9154cde8 |
14-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
resolved conflicts for merge of 3c780188 to master
Change-Id: Ic579d346c27ff05ea6444faaa60fa6caaec86fbf
|
| 236104afb2300b7d969b1dc786f14b5f912df1cf |
14-Mar-2014 |
Glenn Kasten <gkasten@android.com> |
am 089affb6: Merge "mediaserver and associated services are 32-bit only"
* commit '089affb6c2d9ed1e2d49df4be02d30dc490506f9':
mediaserver and associated services are 32-bit only
|
| 3c7801882bbb6d5f3cc641525a54cb8a6c4aca34 |
14-Mar-2014 |
Glenn Kasten <gkasten@android.com> |
am 089affb6: Merge "mediaserver and associated services are 32-bit only"
* commit '089affb6c2d9ed1e2d49df4be02d30dc490506f9':
mediaserver and associated services are 32-bit only
|
| 5b10a2037a835e790994b9ebec3c2e55052f1f3b |
13-Mar-2014 |
Andy Hung <hunga@google.com> |
Fix Offload playback in AudioFlinger
This fixes Offload playback regression introduced by adding
flexible format to mSinkBuffer. Test case is AAC file playback.
Bug: 13450717
Change-Id: I0fa11978295ed4793be90c565e5b8abedf156914
Signed-off-by: Andy Hung <hunga@google.com>
hreads.cpp
|
| 010a1a1a552cdaad362cea8a0333b8906402dbcb |
13-Mar-2014 |
Andy Hung <hunga@google.com> |
Revert "Revert "Convert AudioFlinger mSinkBuffer to flexible format""
This reverts commit e2a9c29f35e0c09782558542fc4cf9823779590e.
hreads.cpp
hreads.h
|
| e2a9c29f35e0c09782558542fc4cf9823779590e |
13-Mar-2014 |
Eric Laurent <elaurent@google.com> |
Revert "Convert AudioFlinger mSinkBuffer to flexible format"
This reverts commit e7e676fd2866fa4898712c4effa9e624e969c182.
Bug: 13450717.
Change-Id: Ib80b0d14428fecce33c62003a1fcf83f71cee03b
hreads.cpp
hreads.h
|
| af783aa91f7a279153fb8bab8d0d6b9f737996e9 |
03-Mar-2014 |
Glenn Kasten <gkasten@android.com> |
mediaserver and associated services are 32-bit only
also 32-bit only command-line apps
Change-Id: I9ac557a8d02bbf6986a9b5c3cdce23d400b306a3
ndroid.mk
|
| c7e5040aa8b8e4da86814bf560346c7571e48087 |
06-Mar-2014 |
Eric Laurent <elaurent@google.com> |
move audio policy service to a separate library
Change-Id: Ibc3ef07aa9860b7fd4f9aaff27b0dbe0dcbf1cbf
ndroid.mk
udioPolicyService.cpp
udioPolicyService.h
|
| e7e676fd2866fa4898712c4effa9e624e969c182 |
05-Mar-2014 |
Andy Hung <hunga@google.com> |
Convert AudioFlinger mSinkBuffer to flexible format
Change-Id: I618d9c99a5f6f8c8d6a9f4b2d19e82c9ddc3b06e
Signed-off-by: Andy Hung <hunga@google.com>
hreads.cpp
hreads.h
|
| 98ef978df4e928f486d244c4d7f7ad9f13111e98 |
04-Mar-2014 |
Andy Hung <hunga@google.com> |
Add mEffectBuffer to AudioFlinger
Change-Id: Icf97c50040bc127723d56eb4d2fb6e790a7253d9
Signed-off-by: Andy Hung <hunga@google.com>
hreads.cpp
hreads.h
|
| f69f9869514730aebe5724c461768507084dfff7 |
07-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
NBAIO::Format_from_SR_C requires audio_format_t parameter
Change-Id: I4b65f6ed2f6ca3608b3a5f88f52a93af0b9b1f4a
hreads.cpp
|
| e857b65c1d3aa055281cb48f59c9b5eb4a062dd0 |
07-Mar-2014 |
Narayan Kamath <narayan@google.com> |
am 3c9130a8: Merge "Make frameworks/av 64-bit compatible"
* commit '3c9130a8a12ce962dc8d22e04243b135b9e98b01':
Make frameworks/av 64-bit compatible
|
| 10ac79930120efea20b125f5faedfe89e8674676 |
07-Mar-2014 |
Glenn Kasten <gkasten@android.com> |
am 5930e8eb: Merge "AudioFlinger: Remove code for supporting resampling in fast tracks"
* commit '5930e8ebdacf3a59c13eef11fd03658ba58b443e':
AudioFlinger: Remove code for supporting resampling in fast tracks
|
| 32fbb48de0248518fc5a14099b32fe4b8e9fc102 |
07-Mar-2014 |
Glenn Kasten <gkasten@android.com> |
am 4361749d: Merge "AudioTrack: Never try to use the fast path if resampling is required"
* commit '4361749d74113069aafb0620a1189404205c58d0':
AudioTrack: Never try to use the fast path if resampling is required
|
| 396a2bc8d1934ccea6159dfcee88e47f0c3c21aa |
07-Mar-2014 |
Glenn Kasten <gkasten@android.com> |
am d66cd743: Merge "audioflinger: conform inline ASM to UAL"
* commit 'd66cd743525cc049033caa3e81cbdf74c72a4556':
audioflinger: conform inline ASM to UAL
|
| 55fbfbbbf00722dac3f789be69138445b771a1e1 |
06-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Use larger capture pipe since we support resampling by 6:1"
|
| 6cbccee701e74fa43a5ea49c15af7dd3267b6699 |
14-Dec-2013 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audioflinger: Fix for a deadlock in track creation
AudioFlinger enters a deadlock (with itself) on trying to free a
RecordTrack or Track object that failed initialization. Clear this
bad object from the caller instead.
Bug: 12423233
Change-Id: I926f2beb922a70f6924e593e2bbf1a5b5df85b16
udioFlinger.cpp
hreads.cpp
|
| 4f4c9a22f4676c65bcae77f0805dcd699b3ba475 |
06-Mar-2014 |
Eric Laurent <elaurent@google.com> |
am 281dd4e1: audioflinger: fix static track end detection
* commit '281dd4e13309973dbb85bce531f884237e0d8fb0':
audioflinger: fix static track end detection
|
| 0c937b4946fae178447d9f8d29200fa2798c05f9 |
05-Mar-2014 |
Eric Laurent <elaurent@google.com> |
am eb16bb5d: am 7028a2c8: am 7e92abea: audioflinger: fix offload write buffer offset
* commit 'eb16bb5dcecf430bfec1c2a4bd15851c83b9c3cc':
audioflinger: fix offload write buffer offset
|
| 7028a2c8106ee5a22db88da5dad581820ee07254 |
05-Mar-2014 |
Eric Laurent <elaurent@google.com> |
am 7e92abea: audioflinger: fix offload write buffer offset
* commit '7e92abeafb184e8a34213d7149592e95a72601b0':
audioflinger: fix offload write buffer offset
|
| 281dd4e13309973dbb85bce531f884237e0d8fb0 |
21-Dec-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix static track end detection
If a static track is not a fast track,
prepareTracks_l() must rely on framesReady() to
detect end of buffer and remove the track from the active
track list.
Failing to do so results in the track staying active but
not processed by the mixer because in underrun. This leaves the
mix buffer content uninitialized and causes the effect process
function to accumulate its output onto undefined data.
Bug: 12013676.
Change-Id: I4b0819a9d93141ac3307b8786fc6a451dd585220
hreads.cpp
|
| abab1c33caf0982c11713e6d64d60105dcbc8ab7 |
06-Jan-2014 |
Gaurav Kumar <gaurav.kumar@broadcom.com> |
AudioMixer: Remove tracks from enabledTracks after reseting outTemp
If any track goes through AudioMixer::process__genericNoResampling, and
its getnextbuffer returns NULL, Then that track is removed by AudioMixer
from enabledTracks.
Thus if all tracks getnextbuffer return NULL, Then this function doesn't
reset outTemp and last buffer in AudioFlinger's mMixBuffer will be
repeated and noise is observed.
Remove tracks from enabledTracks after reseting outTemp to zero, so that
process__genericNoResampling will reset outTemp and noise won't appear.
Bug: 12450065
Change-Id: I28996d425838728955f01eb1a00acf6e6dc2dea1
Signed-off-by: Gaurav Kumar <gaurav.kumar@broadcom.com>
Signed-off-by: Pierre Couillaud <pierre@broadcom.com>
udioMixer.cpp
|
| 7e92abeafb184e8a34213d7149592e95a72601b0 |
22-Nov-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix offload write buffer offset
Fix current audio HAL write buffer offset calculation
which assumes that the frame size is a multiple of 2.
'
Bug: 12823725.
Change-Id: I0195ed5cfef225a6f114e7dd405a02680bb7254e
hreads.cpp
|
| 0504a198d6595e29e58d1d66ae59fac1d662384a |
05-Mar-2014 |
Haynes Mathew George <hgeorge@codeaurora.org> |
am 632c3790: am 7c206e5b: am e0cd1051: audioflinger: update track ready condition
* commit '632c3790523896260dc43fd6cd40a158765cdb15':
audioflinger: update track ready condition
|
| d384a2c44810ce1b51277cc4ab4c5415ff7de6ee |
05-Mar-2014 |
Haynes Mathew George <hgeorge@codeaurora.org> |
am 4f9e38c6: am 21e81bca: am e010f65e: audioflinger: Fix for a deadlock in track creation
* commit '4f9e38c60cc79407a8b2966b3b9780a3e395b017':
audioflinger: Fix for a deadlock in track creation
|
| 7c206e5bd24b5ae4565d0e60b637458a1054dfcf |
05-Mar-2014 |
Haynes Mathew George <hgeorge@codeaurora.org> |
am e0cd1051: audioflinger: update track ready condition
* commit 'e0cd1051ed9fea0629745c29020516ae62298461':
audioflinger: update track ready condition
|
| 21e81bca4d1e6f7d1ccac61b65da7c7a4a121064 |
05-Mar-2014 |
Haynes Mathew George <hgeorge@codeaurora.org> |
am e010f65e: audioflinger: Fix for a deadlock in track creation
* commit 'e010f65e6337267cb15f8894c950a3f64370dd36':
audioflinger: Fix for a deadlock in track creation
|
| 50c3157c5a3e0617be77716beff1ae8801d8a72f |
04-Dec-2013 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audioflinger: check for condition before waiting
AsyncCallbackThread must check for any condition that
was already been satisfied before waiting.
Bug: 11824817
Change-Id: I04683a1f355de4f440106cab47fd916aa39d5e35
hreads.cpp
|
| e0cd1051ed9fea0629745c29020516ae62298461 |
28-Dec-2013 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audioflinger: update track ready condition
Signal track ready if the track isStopping().
Bug: 12423190
Change-Id: I95e14905df10ebf301e398263478c8ca25d7e2ce
racks.cpp
|
| e010f65e6337267cb15f8894c950a3f64370dd36 |
14-Dec-2013 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audioflinger: Fix for a deadlock in track creation
AudioFlinger enters a deadlock (with itself) on trying to free a
RecordTrack or Track object that failed initialization. Clear this
bad object from the caller instead.
Bug: 12423233
Change-Id: I926f2beb922a70f6924e593e2bbf1a5b5df85b16
udioFlinger.cpp
hreads.cpp
|
| 788207057ed4b8df4719ed8089f376ef52de9ca1 |
01-Mar-2014 |
Andy Hung <hunga@google.com> |
Rename mSinkFormat to mMixerFormat for AudioMixer::track_t
AudioMixer::SINK_FORMAT also changes to AudioMixer::MIXER_FORMAT
Change-Id: Ic3f8be77d2c75c082c4fd140bc907e30c304d285
Signed-off-by: Andy Hung <hunga@google.com>
udioMixer.cpp
udioMixer.h
hreads.cpp
|
| 7ed1873acbf43280c5afe51833c3d628fe34e258 |
01-Mar-2014 |
Andy Hung <hunga@google.com> |
Merge "Use float_from_q19_12 conversion in AudioMixer"
|
| 3375bde6c6c1821fca3eeb4840774112e26c9a68 |
01-Mar-2014 |
Andy Hung <hunga@google.com> |
Use float_from_q19_12 conversion in AudioMixer
Change-Id: Idbe84fb9610c7e636a84468f2e2402a913435ca9
Signed-off-by: Andy Hung <hunga@google.com>
udioMixer.cpp
|
| e842614837e5401adf77e90485300c288b9a7876 |
01-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Use larger capture pipe since we support resampling by 6:1
This avoids overruns on the client AudioRecord side,
without requiring client to use a large buffer.
It should not increase input latency, since a newly started AudioRecord
always joins the stream starting at the latest data.
Change-Id: Ib2b8de75cc40a6a3d493a1f8b46b41220f69264f
hreads.cpp
|
| b442921146664fba77b91f221e9b1997ffb8170e |
01-Mar-2014 |
Andy Hung <hunga@google.com> |
Merge "Add MixerBuffer for accumulation of float audio"
|
| 3c35c48a47e9cfbc25778652ff8423f905271a19 |
28-Feb-2014 |
Andy Hung <hunga@google.com> |
Merge "Rename mixBufferSize to mSinkBufferSize in AudioFlinger"
|
| eafbe37da9a08f7599281bc19e000cdf70ae08b6 |
28-Feb-2014 |
Andy Hung <hunga@google.com> |
Merge "Rename constants for MixBufferSize to SinkBufferSize"
|
| 625b57c5ab5fefde1547d89a061420be733fb7dd |
28-Feb-2014 |
Andy Hung <hunga@google.com> |
Merge "Rename mMixBuffer to mSinkBuffer in AudioFlinger"
|
| d2304db2fcb5112292105a0949a55986a4c9875f |
03-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Rename setStreamOutput to invalidateStream
And simplify by removing the unused I/O handle parameter 'output'.
Change-Id: Ie9c4df17a7378066312d4ed8790fda7a9125c95e
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
|
| 69aed5f0f4a3be3996d1e78a0473e1a72c1547da |
26-Feb-2014 |
Andy Hung <hunga@google.com> |
Add MixerBuffer for accumulation of float audio
Path into and out of MixerBuffer is set up, but not
currently enabled.
Change-Id: I9d50752607d22dd2a3d9cc7e053babf8dfb22958
Signed-off-by: Andy Hung <hunga@google.com>
hreads.cpp
hreads.h
|
| 25c2dac12114699e90deb1c579cadebce7b91a97 |
27-Feb-2014 |
Andy Hung <hunga@google.com> |
Rename mixBufferSize to mSinkBufferSize in AudioFlinger
Change-Id: I4d1d18d716fa3ad8c0e7cc87f5a2823422f504eb
Signed-off-by: Andy Hung <hunga@google.com>
hreads.cpp
hreads.h
|
| 09a5007b17acb49d25cfa386a2e2534d942e8854 |
27-Feb-2014 |
Andy Hung <hunga@google.com> |
Rename constants for MixBufferSize to SinkBufferSize
Updated variable names are
kMinNormalSinkBufferSizeMs
kMaxNormalSinkBufferSizeMs
Change-Id: I91969eae3f8c10275db8778634cbf05fdec0712b
Signed-off-by: Andy Hung <hunga@google.com>
hreads.cpp
|
| 2098f2744cedf2dc3fa36f608aa965a34602e7c0 |
27-Feb-2014 |
Andy Hung <hunga@google.com> |
Rename mMixBuffer to mSinkBuffer in AudioFlinger
Change-Id: I1f55630425b8ce36e9df2aaf64e06a8960b69a47
Signed-off-by: Andy Hung <hunga@google.com>
hreads.cpp
hreads.h
|
| 8897759002677a7072a51a7f0b7cc63761c5f2d4 |
27-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Move up AudioFlinger::createTrack and openRecord parameter checks"
|
| 53b5d098b7d5d4412b9b9fe08daaf84c56b716e9 |
05-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Move up AudioFlinger::createTrack and openRecord parameter checks
Perform the basic sample rate, format, and channel masks tests as
early as possible. Now the thread methods only need to do the thread
type-specific checks.
Change-Id: I57d8319a228dda0446ca9b3ddcb7c9f67f28238c
udioFlinger.cpp
|
| 4d3c5bde85c5404d7927a770da90a5c9d09fb6a7 |
27-Feb-2014 |
Andy Hung <hunga@google.com> |
Merge "Add Track Sink Format to AudioMixer"
|
| a1ab7cc8611c83427b57f6d4d4ce7aad9d1c0330 |
25-Feb-2014 |
Andy Hung <hunga@google.com> |
Add Track Sink Format to AudioMixer
Track Sink Format now allows both AUDIO_FORMAT_PCM_FLOAT and
AUDIO_FORMAT_PCM_16_BIT (default). The float case is not
enabled now.
Change-Id: Ibee70cab4725c8bc9905e49f1f9a9c2448f76e00
Signed-off-by: Andy Hung <hunga@google.com>
udioMixer.cpp
udioMixer.h
|
| b78d804b1090a430a126a6c356037ded96ffd91a |
25-Feb-2014 |
Andy Hung <hunga@google.com> |
Merge "Fix 1:1 dynamic resampler rate setting"
|
| 9d518f1aa50da9c3dfadb97090fb79aac886da12 |
25-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "The -O option now takes a CSV instead of a single value"
|
| 323da1015a758243c2c43017b026a01c6cf0c6f2 |
22-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Account for unreleased frames when predicting resampler needs"
|
| d0e0cfa58a35508c14818b88804845194b5d80e1 |
22-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Make tee sink work again"
|
| 607fa3e928de696eba49f198af72d68e4591ca1b |
21-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Account for unreleased frames when predicting resampler needs
Other changes related to resampling:
- improve logs
- use correct form of assert
- don't round up max usable circular buffer sizes to power of 2;
this gives more predictable results
- add some FIXMEs
Bug: 13133176
Change-Id: I8d3c90eed4eb0c0c1fbac3c864e179a2806c2301
hreads.cpp
|
| c52b033f0be38ca915d389b672bcf9c46b4c78ef |
21-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
The -O option now takes a CSV instead of a single value
Also added -P option to provide a CSV list of provided values.
Change-Id: I4e0d6fe8b062d27d95d33493f80bd8f1c510d2c8
est-resample.cpp
|
| f66b42242342017c26eb97de544dae31dd2537ca |
20-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Make tee sink work again
It was broken by this earlier change to NBAIO:
> Change-Id: I5eda412648b094358f5eefc38300e9ec8a734cd3
But the code was not being compiled, so the error was not caught earlier.
Also increase the default size of per-track pipe to a reasonable value.
Change-Id: Ica05017e6c6533e1fea9df379a9b204eebed4a1f
udioFlinger.cpp
udioFlinger.h
|
| 8ea16e4b0a7d398d26887c18675b3899de5d779d |
21-Feb-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix race condition in SyncEvent callback
Now that the SyncEvent callback is implemented by the
RecordTrack instead of the RecordThread, there is a possibility
that the callback is called after the track deletion.
SyncEvent callback now uses a weak pointer instead of
a raw pointer as cookie. This allows the callback implementer to
acquire a strong reference on the object pointed to by the cookie.
Bug: 13114128.
Change-Id: Id61b8f06044ed1e52c6f7e7c666cdede68340de2
udioFlinger.cpp
udioFlinger.h
hreads.cpp
|
| 1af34085e18c4d5ab297232f167a71e89ff7f65d |
20-Feb-2014 |
Andy Hung <hunga@google.com> |
Fix 1:1 dynamic resampler rate setting
Change-Id: Id69e98ebaaca8dd8c976eb823f1cc1024b0eddf5
Signed-off-by: Andy Hung <hunga@google.com>
udioResamplerDyn.cpp
|
| 56df9ff31d583ad3eae4f279a3df550273c58e1e |
17-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Allow specifying frames per call to resample()
Bug: 13073201
Change-Id: Id2f0fcd7562d9ba5a58c128d71bbba42dfea86cc
est-resample.cpp
|
| d20db4c4d8eb2593b5d0214890420a49e0b01796 |
20-Feb-2014 |
Andy Hung <hunga@google.com> |
Merge "Fix resampler to allow output of single frame"
|
| 24781fff62a4cf7279d3dac83c33e2ac612712ba |
19-Feb-2014 |
Andy Hung <hunga@google.com> |
Fix resampler to allow output of single frame
Bug: 13073201
Change-Id: If7818a3389a191a37277bbd8e96a59ef8ce68509
Signed-off-by: Andy Hung <hunga@google.com>
udioResampler.cpp
udioResampler.h
udioResamplerCubic.cpp
udioResamplerDyn.cpp
udioResamplerSinc.cpp
|
| 4cc0a6a835c806d200ef83ef31fe5bef327c355c |
17-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Multi-client recording continued
Remove mReqChannelCount and mReqSampleRate.
Increase number of input frames needed for sample rate converter.
Add TODO comments to investigate later.
Fix bugs in sample rate conversion.
Change-Id: I3aa54b862843d518b6e4ece9a7f52fd27d741629
udioFlinger.cpp
hreads.cpp
hreads.h
|
| 25f4aa83efaa9179e65a20583a6d441de2c3ff3f |
07-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Move handleSyncStartEvent and clearSyncStartEvent to RecordTrack
Change-Id: I611a63e63acf3eb17e2072c1a77b91fe9e62e36a
ecordTracks.h
hreads.cpp
hreads.h
racks.cpp
|
| deca2ae0a7cf8bc54ff3f30b7dc39bbc78b94c0d |
07-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Add _l suffix to readInputParameters/OutputParameters
They are always called while thread mutex is locked,
or in the thread constructor.
Change-Id: I56df71563a52a5cd6cea6fc0914df993519e1b74
ecordTracks.h
hreads.cpp
hreads.h
|
| 6dd62fb91d82dedcfa3ab38c02eb0940b4ba932a |
06-Dec-2013 |
Glenn Kasten <gkasten@google.com> |
Multi-client recording
Supports multiple clients both at native sample rate and with resampling.
Change-Id: Icea55b4fd30751761b7debaa3ce016c79e712d8d
ecordTracks.h
hreads.cpp
hreads.h
rackBase.h
racks.cpp
|
| c940c38242e8e0265db25a646ad63263934f2bb9 |
13-Feb-2014 |
Marco Nelissen <marcone@google.com> |
Merge "Track pid for each session"
|
| c6d7ab74962bfd24d105277b28248b4819763abd |
12-Feb-2014 |
Andy Hung <hunga@google.com> |
Merge "Proper adjustment of filterlength for downsampling"
|
| a3bb9a3a25b58669d75da4f73764ac4c2bf34158 |
11-Feb-2014 |
Andy Hung <hunga@google.com> |
Proper adjustment of filterlength for downsampling
Bug: 12963040
Change-Id: I772b8efd3f7eae6d075994b210dd0168f5365f4d
Signed-off-by: Andy Hung <hunga@google.com>
udioResamplerDyn.cpp
|
| d457c970c8d08519cd77280a90b61ae1e342cfe3 |
11-Feb-2014 |
Marco Nelissen <marcone@google.com> |
Track pid for each session
so they can be properly freed.
Change-Id: I6f389035bc29e74e7c367c1c6d0252b180f666b3
udioFlinger.cpp
udioFlinger.h
|
| 1d6fa7af1288b550faabe4ec2cf98684236723db |
11-Feb-2014 |
Narayan Kamath <narayan@google.com> |
resolved conflicts for merge of 566be7c3 to master
Change-Id: I7b1cc71057b2bd4f771e7bcf508a8c3abd6017ce
|
| 377b2ec9a2885f9b6405b07ba900a9e3f4349c38 |
03-Feb-2014 |
Kévin PETIT <kevin.petit@arm.com> |
Make frameworks/av 64-bit compatible
Contains the necessary changes to make frameworks/av build and work
on a 64-bit machine.
Signed-off-by: Craig Barber <craig.barber@arm.com>
Signed-off-by: Kévin PETIT <kevin.petit@arm.com>
Signed-off-by: Ashok Bhat <ashok.bhat@arm.com>
Signed-off-by: Marcus Oakland <marcus.oakland@arm.com>
Change-Id: I725feaae50ed8eee25ca2c947cf15aee1f395c43
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
ffects.cpp
astMixer.cpp
tateQueue.cpp
hreads.cpp
hreads.h
racks.cpp
|
| b1f5b0dd237c2767ad7bc0b081d03aafc87589ea |
10-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix clang warnings in AudioFlinger"
|
| 4b76d27d6c4751b31a1cb8ac5e6da1d4b7724a7b |
10-Feb-2014 |
Glenn Kasten <gkasten@android.com> |
am 5a458ede: am 5930e8eb: Merge "AudioFlinger: Remove code for supporting resampling in fast tracks"
* commit '5a458ede3b7d303723fde4b856532bdc3b94971b':
AudioFlinger: Remove code for supporting resampling in fast tracks
|
| 3523e8c40bc60af0c95d1aa71a51a13d69ec80a2 |
10-Feb-2014 |
Glenn Kasten <gkasten@android.com> |
am 4e12dade: am 4361749d: Merge "AudioTrack: Never try to use the fast path if resampling is required"
* commit '4e12dadeec10993d2f77cce96600e73ca0dd86e2':
AudioTrack: Never try to use the fast path if resampling is required
|
| 5930e8ebdacf3a59c13eef11fd03658ba58b443e |
10-Feb-2014 |
Glenn Kasten <gkasten@android.com> |
Merge "AudioFlinger: Remove code for supporting resampling in fast tracks"
|
| 4361749d74113069aafb0620a1189404205c58d0 |
10-Feb-2014 |
Glenn Kasten <gkasten@android.com> |
Merge "AudioTrack: Never try to use the fast path if resampling is required"
|
| 9601c6efcb2552960d6f125d073525b581c1b7ec |
10-Feb-2014 |
Marco Nelissen <marcone@google.com> |
Merge "Pretty up audioflinger dumpsys"
|
| 01d3acba9de861cb2b718338e787cff3566fc5ec |
06-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Fix clang warnings in AudioFlinger
Change-Id: I0fa61025c979709ad7d655bc717df5f194b6089e
udioFlinger.h
udioMixer.h
udioPolicyService.cpp
udioResampler.cpp
udioResampler.h
ffects.cpp
ffects.h
|
| ec95f0e82a680931e9c7c9011dbbd68d5ab484c8 |
10-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Use printf format %#x for audio_format_t in logs"
|
| b220884bf3129253cc5bc8d030bc475411ea4911 |
07-Feb-2014 |
Marco Nelissen <marcone@google.com> |
Pretty up audioflinger dumpsys
Change-Id: I57e44b4c36b99f7149542bbcf9645521c6152dfa
udioFlinger.cpp
udioFlinger.h
ffects.cpp
astMixer.cpp
laybackTracks.h
ecordTracks.h
hreads.cpp
racks.cpp
|
| 8f762d577f2dcf34357eb8731070ae465649ec9a |
08-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix whitespace"
|
| a793625031a3e326a261a0843598c35a5784bbde |
07-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Use Format_isValid() and Format_isEqual() instead of direct comparison"
|
| 4944acb7355b3aa25748fd25945a363a69d65444 |
19-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Fix whitespace
Change-Id: I9add0872030a26933f6b6dad1be299154611cc56
udioPolicyService.cpp
hreads.cpp
hreads.h
racks.cpp
|
| cac3daa6332bf6d1f7d26adc4a9915f3d7992dd9 |
07-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Use printf format %#x for audio_format_t in logs
Change-Id: I1c611d1037685d52ccc84efe0fccd6413ec938e9
udioFlinger.cpp
hreads.cpp
|
| f74f213c0cd0612101e00a73704429dac1a92fe3 |
07-Feb-2014 |
Eric Laurent <elaurent@google.com> |
Merge "AudioFlinger: Offload track transition fix"
|
| d9a4783985203c759ddcca784798ef395668fe05 |
07-Feb-2014 |
Eric Laurent <elaurent@google.com> |
Merge "AudioFlinger: Modify flush handling for offload path"
|
| c9583cba9913682e1c181675c6d0066292fb719d |
07-Feb-2014 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: OffloadThread state fix"
|
| 010c4c8ec50eb04ece15f792c508a3b869749310 |
07-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix IoConfigEvent constructor initializer list"
|
| 4c6a433d74d5ae8b9bc0557207e3ced43bf34a25 |
15-Jan-2014 |
Haynes Mathew George <hgeorge@codeaurora.org> |
AudioFlinger: Offload track transition fix
Trigger state changes in OffloadThread on adding a new offload track.
Bug: 12530661
Change-Id: I95b62764c7ac041d470702e86b951f812eb0e70f
hreads.cpp
hreads.h
|
| 7844f679be8d94c5cdf017f53754cb68ee2f00da |
15-Jan-2014 |
Haynes Mathew George <hgeorge@codeaurora.org> |
AudioFlinger: Modify flush handling for offload path
Do not allow an offload track to directly control the offload thread
behavior. OffloadThread can check for any pending flush reporting
by its active tracks and decide to flush the HW or not.
Bug: 12530661
Change-Id: Ib33f023c942f6c091b618004136b153c38a6eef6
laybackTracks.h
hreads.cpp
hreads.h
racks.cpp
|
| 0f02f265123b7ef2fd6ac09ff70cde26eb5559ad |
11-Jan-2014 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audioflinger: OffloadThread state fix
When switching from one offload session to another that
reuses the same offload output, audio HAL is paused & flushed in
preparation for the new track
OffloadThread must reset mHwPaused to allow proper state transitions
for the second track. In addition, Offload thread need not
explicitly call resume on HAL while handling a flush.
Bug: 12530661
Change-Id: I8173845e1782b5a8b372a82c7d337299a61e0efe
hreads.cpp
|
| 3ce28aa3cb5262775180a8b423cfb4a5670ebc59 |
05-Feb-2014 |
Martin Storsjo <martin@martin.st> |
AudioFlinger: Remove code for supporting resampling in fast tracks
This isn't used at the moment.
Change-Id: I4e0fb2af5f7d959dbafd5ddb7defa1c6b8e8636a
astMixer.cpp
astMixerState.cpp
astMixerState.h
hreads.cpp
|
| 4a25471795b05ffcfcc079e9f9cb5ed549023d14 |
05-Feb-2014 |
Glenn Kasten <gkasten@android.com> |
am 5ef74dce: am d66cd743: Merge "audioflinger: conform inline ASM to UAL"
* commit '5ef74dce64a1b70534547c044a6d15533f4a9dca':
audioflinger: conform inline ASM to UAL
|
| 7332c37c459f5f382a9e4105d522d1c5a7f6f0b5 |
05-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "audioflinger: Fix for a deadlock in track creation"
|
| 2a16afb8dc056877f34124ba5befe75171de930f |
05-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix unused parameter warnings in audio"
|
| 5f51ade2c290e239a125dc88943b240e1105fd97 |
04-Feb-2014 |
synergy dev <synergye@codefi.re> |
audioflinger: conform inline ASM to UAL
Clang requires some inline ASM to conform to the UAL standards (Unified Assembler Language).
This fixes a small issue in this inline asm to allow building.
Change-Id: Ifd9b1814343ab5ade636b9401a21d575559dac16
udioResampler.cpp
|
| 6e0d67d7b496ce17c0970a4ffd3a6f808860949c |
31-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Use Format_isValid() and Format_isEqual() instead of direct comparison
Change-Id: Ie87607aa514976947540a77775e6425c4e56e7d9
udioFlinger.cpp
astMixer.cpp
racks.cpp
|
| a98a58ce32f4ad1bdb7692184eb17622fb11e397 |
04-Feb-2014 |
Andy Hung <hunga@google.com> |
Merge "Fix dynamic resampling for non-NEON devices"
|
| 83be2560d9396b3bd32919123bd67a783e6aaf7c |
03-Feb-2014 |
Andy Hung <hunga@google.com> |
Fix dynamic resampling for non-NEON devices
The stride by 16 code template automatically falls back to stride by 2
(as the NEON code template override does not occur).
As such the stride by 2 (in the switch) was removed as it was unnecessary.
Removing the if() causes the fall back to happen properly for non-NEON
compilation.
Change-Id: Ia7aec6ae1d88eb54377e56e2fb57f2ae5b74cf1b
Signed-off-by: Andy Hung <hunga@google.com>
udioResamplerDyn.cpp
|
| 0f11b51a57bc9062c4fe8af73747319cedabc5d6 |
01-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Fix unused parameter warnings in audio
Change-Id: I665ba3358dd9502f0adec70d486e7bf8a2e1b0fe
udioFlinger.cpp
udioPolicyService.cpp
ffects.cpp
hreads.cpp
hreads.h
racks.cpp
|
| 49debee5bdddd7a1e0742a3ab66d4630bbe25ca2 |
01-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Fix IoConfigEvent constructor initializer list
Fortunately this bug never showed up, because the parameter
field has not been used yet.
Change-Id: I2b36e2b7e1dcc2039c494fc5ea4994d262435aac
hreads.h
|
| 398f21348e5100289f6e5be30c8b5257fa04aaf9 |
31-Jan-2014 |
Martin Storsjo <martin@martin.st> |
AudioTrack: Never try to use the fast path if resampling is required
Unless AudioFlinger was built with FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE
enabled, AudioFlinger would deny using the fast path (and internally
fall back to the normal codepath) when it realized that resampling
was required. Since the buffer size calculations within AudioFlinger
don't take resampling into account properly (see the calculation
below "AUDIO_OUTPUT_FLAG_FAST denied" in audioflinger/Threads.cpp,
just below the hunk that this patch changes), make sure AudioTrack
doesn't try to use the fast path if resampling is required.
This removes the possibility to enable
FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE in AudioFlinger since it
AudioTrack now won't even try to use the fast path for content
that requires resampling, regardless of the AudioFlinger configuration.
Change-Id: Icf0f8ad50bf0fdb84657f518c0120aa0535f23f9
onfiguration.h
hreads.cpp
|
| 9e0308c03d4e76d3146cbb6e30aeb3ac03f05cf5 |
30-Jan-2014 |
Andy Hung <hunga@google.com> |
Enable new dynamic audio resampler as default
The default may still be overridden by af.resampler.quality
in the local or build properties file. For the build properties
check the relevant device.mk files.
In the absence of being overridden, the 44.1 to 48kHz and
48 to 44.1kHz conversion will use the DYN_MED_QUALITY resamplers
and other conversion will use the DYN_LOW_QUALITY resamplers.
Change-Id: Icf9a4af5422cddc445e1904eef93984b0abe6a44
Signed-off-by: Andy Hung <hunga@google.com>
udioMixer.cpp
udioResampler.cpp
|
| 745e9a8283b56c7772ee7d72383a3f2e012e1ef9 |
21-Dec-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix static track end detection
If a static track is not a fast track,
prepareTracks_l() must rely on framesReady() to
detect end of buffer and remove the track from the active
track list.
Failing to do so results in the track staying active but
not processed by the mixer because in underrun. This leaves the
mix buffer content uninitialized and causes the effect process
function to accumulate its output onto undefined data.
Bug: 12013676.
Change-Id: Iad72c921fa18d34811abf7d1073890c093a27725
hreads.cpp
|
| e983e0a0017fce81dc3d9bea36f5abb7b7bce40b |
30-Jan-2014 |
Andy Hung <hunga@google.com> |
Merge "Remove warnings in Audio mixer resampler compilation."
|
| 03e9e83c47ab4a518da0a1f36b8f702f59221c95 |
14-Dec-2013 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audioflinger: Fix for a deadlock in track creation
AudioFlinger enters a deadlock (with itself) on trying to free a
RecordTrack or Track object that failed initialization. Clear this
bad object from the caller instead.
Bug: 12423233
Change-Id: I926f2beb922a70f6924e593e2bbf1a5b5df85b16
udioFlinger.cpp
hreads.cpp
|
| ee931ff7d6620e5705f4dfba901fdb03fa4a35fd |
28-Jan-2014 |
Andy Hung <hunga@google.com> |
Remove warnings in Audio mixer resampler compilation.
Change-Id: Ie68766142796772a54a90afdea188d0a5a78d4ac
Signed-off-by: Andy Hung <hunga@google.com>
udioMixer.cpp
udioResampler.cpp
|
| 000a4193dd82549192277fd4b9bb571d8a4c262f |
30-Jan-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix record thread exit pending check
RecordThread loop must not release the mutex after checking for
exitPending and before waiting for a new wake up condition.
This can happen under the hood when methods like processConfigEvents_l()
or checkForNewParameters_l() are called. So exitPending() must
be checked after calling these functions.
Bug: 12787961.
Change-Id: Ia18c518bd5344fbb2401067303fcfe76a86879c4
hreads.cpp
|
| 0eed5ac1a14067d0fd51282e97b43b4fd9121842 |
29-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Merge commit 'adf3f9b9' into manualmerge
Change-Id: Icdef7d075b160eb2a5ced7c57f90c6a6a4b4c64a
|
| 1bf0eecd2ec6a3b3e29ad2c91bcefda76abef2fe |
29-Jan-2014 |
Glenn Kasten <gkasten@android.com> |
Merge "AudioMixer: Remove tracks from enabledTracks after reseting outTemp"
|
| 7e79cd2d3067cdb7b9daf639cf772d4dd149c934 |
06-Jan-2014 |
Gaurav Kumar <gaurav.kumar@broadcom.com> |
AudioMixer: Remove tracks from enabledTracks after reseting outTemp
If any track goes through AudioMixer::process__genericNoResampling, and
its getnextbuffer returns NULL, Then that track is removed by AudioMixer
from enabledTracks.
Thus if all tracks getnextbuffer return NULL, Then this function doesn't
reset outTemp and last buffer in AudioFlinger's mMixBuffer will be
repeated and noise is observed.
Remove tracks from enabledTracks after reseting outTemp to zero, so that
process__genericNoResampling will reset outTemp and noise won't appear.
Bug: 12450065
Change-Id: I1ccac7ee4a3bf7fd930254356c072099e11e7c19
Signed-off-by: Gaurav Kumar <gaurav.kumar@broadcom.com>
Signed-off-by: Pierre Couillaud <pierre@broadcom.com>
udioMixer.cpp
|
| f0002d142e6d24c5438600b2c259679de710f8ac |
24-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Replace control block frameCount_ by explicit in/out parameter"
|
| 0ce5ca5e113d58cbcb7cb48615d6dbbf3fbe7c66 |
17-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Cleanup createTrack error handling"
|
| 6a507298357c9ff869b97db57a890e58bd6bdec0 |
17-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioRecord::getInputFramesLost() cleanup"
|
| 0cde076ddb283c84c3801a2df4cc3df99bd1577f |
17-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Cleanup createTrack error handling
Similar to cleanup done earlier for openRecord in this CL:
> Change-Id: I84897dd7d30b370640b54e928f230604b873cb68
Bug: 10888816
Change-Id: I804a47f898e0319a60a9bd58906bbb037e45cc25
udioFlinger.cpp
hreads.cpp
|
| dbed8955148dd2fe39584007c528e3a532f1a04e |
15-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix error handling in AudioSystem::getOutputForEffect"
|
| 5f972c031d4061f4f037c9fda1ea4bd9b6a756cd |
13-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
AudioRecord::getInputFramesLost() cleanup
Fixed bug that if the binder call failed (for example if the
IAudioFlinger binder is dead), then getInputFramesLost was returning
garbage. Now it correctly returns zero, which is the error value for
this method.
The type declarations for getInputFramesLost were inconsistent:
a mixture of unsigned int, size_t, and uint32_t. Now it returns uint32_t
everywhere, which is what the underlying HAL API returns.
Added a FIXME about the side effect behavior. This will need review
for multi-client.
Change-Id: Ifa2e117a87dbd0c1f2c892a31d1c3dd919bf1a0a
udioFlinger.cpp
udioFlinger.h
hreads.cpp
hreads.h
|
| 5b27ccd67c845aa20a12a1fb58339e7e81e3d536 |
15-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Use AUDIO_SESSION_ALLOCATE instead of 0"
|
| 03f696e3eb89c0642f9a2b761ff2cfc6af003d16 |
10-Jan-2014 |
Andy Hung <hunga@google.com> |
Merge "Improve dynamic audio resampler filter generation"
|
| 6340829ffb9ff7586ffb88a214f56340d4c70d60 |
10-Jan-2014 |
Glenn Kasten <gkasten@android.com> |
am 0ba9f9b0: am d8d45edf: am 61540b5f: Merge "Frameworks: AudioFlinger: Fix effects memory leak"
* commit '0ba9f9b07d47e9e6b374798f397fe961256a6029':
Frameworks: AudioFlinger: Fix effects memory leak
|
| 61540b5fe82cad1c6baa018f02bd2554e62e4364 |
10-Jan-2014 |
Glenn Kasten <gkasten@android.com> |
Merge "Frameworks: AudioFlinger: Fix effects memory leak"
|
| efa6ea97022780b68e595e5326e30dbe2a799202 |
08-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Fix error handling in AudioSystem::getOutputForEffect
and AudioPolicyService::getOutputForEffect.
The conventional error value for audio_io_handle_t is 0,
not a status_t cast to audio_io_handle_t.
Change-Id: I34b3fd1a50f3fa1cbf39f32eea1911112a4e094a
udioPolicyService.cpp
|
| 6582f2b14a21e630654c5522ef9ad64e80d5058d |
03-Jan-2014 |
Andy Hung <hunga@google.com> |
Improve dynamic audio resampler filter generation
Improve dynamic audio resampler filter generation speed by 2x.
The resulting filters should be the same (excepting roundoff).
Also added check for upsampling sample rate changes to share
previously generated filters.
Modify the profiling to improve output format and sampling result
reliability.
Change-Id: I9aa6b914fd552a63f79dd4a95945df2f8275772a
Signed-off-by: Andy Hung <hunga@google.com>
udioResamplerDyn.cpp
udioResamplerDyn.h
udioResamplerFirGen.h
est-resample.cpp
|
| 0bcfa88149e2404b34d13c622e3921e1b846cdf8 |
28-Dec-2013 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audioflinger: update track ready condition
Signal track ready if the track isStopping().
Bug: 12423190
Change-Id: Ie279995d5f90fa8211a20dfbeacc1cf9b921d0bf
racks.cpp
|
| 67d15f9ab45e7c03b23c871f680b1fae577ac43a |
07-Jan-2014 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: fix static track end detection"
|
| 86eae0e5931103e040ac2cdd023ef5db252e09f6 |
09-Dec-2013 |
Andy Hung <hunga@google.com> |
Audio resampler update to add S16 filters
This does not affect the existing resamplers.
New resampler accessed through additional quality settings:
DYN_LOW_QUALITY = 5
DYN_MED_QUALITY = 6
DYN_HIGH_QUALITY = 7
Change-Id: Iebbd31871e808a4a6dee3f3abfd7e9dcf77c48e1
Signed-off-by: Andy Hung <hunga@google.com>
ndroid.mk
udioResampler.cpp
udioResampler.h
udioResamplerDyn.cpp
udioResamplerDyn.h
udioResamplerFirGen.h
udioResamplerFirOps.h
udioResamplerFirProcess.h
udioResamplerFirProcessNeon.h
est-resample.cpp
|
| 13e4c960ea3db03a43e084fbd85d52aa77f7b871 |
21-Dec-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix static track end detection
If a static track is not a fast track,
prepareTracks_l() must rely on framesReady() to
detect end of buffer and remove the track from the active
track list.
Failing to do so results in the track staying active but
not processed by the mixer because in underrun. This leaves the
mix buffer content uninitialized and causes the effect process
function to accumulate its output onto undefined data.
Bug: 12013676.
Change-Id: Iad72c921fa18d34811abf7d1073890c093a27725
hreads.cpp
|
| aea7ea06394bcb155972d82055d4ea59962e4051 |
26-Jun-2013 |
Glenn Kasten <gkasten@google.com> |
Use AUDIO_SESSION_ALLOCATE instead of 0
Also fix a couple of places where we were using AUDIO_SESSION_OUTPUT_MIX,
which happens to also be equal to 0, but has a different meaning.
Change-Id: I90e39be3b89f5021a96d9e3b8d10929013ca977f
udioFlinger.cpp
|
| 74935e44734c1ec235c2b6677db3e0dbefa5ddb8 |
19-Dec-2013 |
Glenn Kasten <gkasten@google.com> |
Replace control block frameCount_ by explicit in/out parameter
in IAudioFlinger::createTrack and IAudioFlinger::openRecord
Change-Id: I09c644c80e92c8e744b1b99055988a2588b2a83d
udioFlinger.cpp
udioFlinger.h
hreads.cpp
hreads.h
racks.cpp
|
| 47f3f5a119194d4c06815453be9950ba112c8e3e |
18-Dec-2013 |
Glenn Kasten <gkasten@google.com> |
Fix bug in test-resample's AudioBufferProvider
The contract for AudioBufferProvider::releaseBuffer() was missing.
Bug: 12194314
Change-Id: I2fcf75e7b8eaf6db34f360206d79457a04a73565
est-resample.cpp
|
| bd72d22097f78f5bd668b223bc8c94e351311e31 |
18-Dec-2013 |
Glenn Kasten <gkasten@google.com> |
Add ability to read .wav files to test-resample
Previously test-resample could only read .raw files, and the input
sample rate had to be specified. Now the input sample rate is derived
from the input file. This also allows us to read 8-bit PCM files,
and other formats such as floating-point in the future.
However, the ability to read raw files is lost.
A workaround is to use sox or equivalent on the host.
Change-Id: Icd06b4d02482b3ad07bf03979f46860e68d38ad9
est-resample.cpp
|
| f5293648b727fb3909cd2300a73377f032f8b050 |
17-Dec-2013 |
Glenn Kasten <gkasten@google.com> |
Use libsndfile to write .wav files
This will reduce code duplication, and allow us take advantage of more
advanced capabilities of libsndfile in the future.
Change-Id: I25fa2b6d0c21e325aeaf05bda62cf7aab0c5deb4
ndroid.mk
est-resample.cpp
|
| 9a1c785c019bdb25fefa7d1c0e50862ddc0e4e21 |
18-Dec-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix several test-resample BufferProvider bugs"
|
| e00eefe64e3bad166c672db96c9c35992766e819 |
17-Dec-2013 |
Glenn Kasten <gkasten@google.com> |
Fix several test-resample BufferProvider bugs
Previously getNextBuffer always returned the same data address over
and over. Now it correctly returns the right portion of the input buffer.
Previously getNextBuffer always returned the total number of frames in
the input, which might be larger than the size requested by the caller,
and/or larger than the number of remaining input frames. It also always
returned successfully, even when there should be no frames available.
This violates the contract for getNextBuffer. Now getNextBuffer will
return the maximum of the number of frames requested, and the number of
remaining frames available. If that maximum is zero, getNextBuffer will
return an error instead.
Previously releaseBuffer would silently allow releasing more frames than
were actually gotten, which violates the contract for releaseBuffer.
Now releaseBuffer checks for this and logs a message if it happens.
Add 'v' (verbose) option to log buffer provider calls.
Bug: 12194314
Change-Id: I9b915e954b3612a07ef271da8652486b8875e0fd
est-resample.cpp
|
| c95616357a911e5c1c55873492b05ebfa3cf1894 |
04-Dec-2013 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audioflinger: check for condition before waiting
AsyncCallbackThread must check for any condition that
has already been satisfied before waiting.
Bug: 11824817
Change-Id: Ic8c2090d521ecd6a30b76ee75635258d35eb1eff
hreads.cpp
|
| 688ca6330a5e7d072f67a1791e464fa7dec08b72 |
17-Dec-2013 |
Glenn Kasten <gkasten@google.com> |
am 5c474238: am 5893736d: am a67c7653: Merge "Increase kFastTrackMultiplier from 1 to 2"
* commit '5c474238bc579224ec8f7267d05651b0aa934100':
Increase kFastTrackMultiplier from 1 to 2
|
| a67c76538fff102cd44b2f38dd073df5f2a33605 |
17-Dec-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Increase kFastTrackMultiplier from 1 to 2"
|
| 24a325d6f8c4bbf9330e6ce0c769d46e04266ffc |
04-Dec-2013 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audioflinger: check for condition before waiting
AsyncCallbackThread must check for any condition that
was already been satisfied before waiting.
Bug: 11824817
Change-Id: Ic8c2090d521ecd6a30b76ee75635258d35eb1eff
hreads.cpp
|
| 70aca464f7e39a3b208341f442b6fc2316e17147 |
05-Dec-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Add RecordThread media.log and deferred deallocation"
|
| 0312b69e65e4c8bf6a0c0757d9100dee3651c16e |
05-Dec-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Change mActiveTrack to mActiveTracks vector"
|
| 481fb67a595f23c5b7f5be84b06db9b84a41a42f |
30-Sep-2013 |
Glenn Kasten <gkasten@google.com> |
Add RecordThread media.log and deferred deallocation
This change allows a media.log buffer for RecordThread.
Unlike playback threads which stick around forever, the RecordThread comes
and goes for every capture session. This means that the media.log buffer
for a RecordThread would disappear too, and so was useless. Now when a
thread exits, it's associated media.log buffer is just marked for deferred
deallocation. It is only actually freed when the memory is needed.
Other changes:
- Fix bug in unregistering comparison, it was comparing the wrong pointers
- Increased size of log area so we can log for RecordThread also
Change-Id: If45d4c03a793b86390a0112ec3acc5d41b2e3635
udioFlinger.cpp
udioFlinger.h
hreads.cpp
|
| b5fed68bcdd6f44424c9e4d12bfe9a3ff51bd62e |
03-Dec-2013 |
Glenn Kasten <gkasten@google.com> |
Increase kFastTrackMultiplier from 1 to 2
Change-Id: I158f147295eebcea96e4047d7618069bc48bdd7d
hreads.cpp
|
| 866d90595a177f6027e97161781948baebf2aa68 |
02-Dec-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Code simplification and movement to prepare for mActiveTracks"
|
| 480acec40aa2d96119530c7d451282af8cf07327 |
23-Nov-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix bug in creation of EffectHandle when out of memory"
|
| 639512ce33fb19c876b1be68555d83ccda6e7ad7 |
23-Nov-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Consistent error checking for sp<IMemory> and pointer()"
|
| 04733dbdede313fc5e291070014c1a0641fe6166 |
22-Nov-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix offload write buffer offset
Fix current audio HAL write buffer offset calculation
which assumes that the frame size is a multiple of 2.
Change-Id: I0195ed5cfef225a6f114e7dd405a02680bb7254e
hreads.cpp
|
| 2b806406ac1ec680b6fe3aaa84c54bdc4e43ad8d |
21-Nov-2013 |
Glenn Kasten <gkasten@google.com> |
Change mActiveTrack to mActiveTracks vector
The size of mActiveTracks is still limited to 1 for now.
Change-Id: Ia6b06b25d8c4709f087110ae29c589999407a2a2
hreads.cpp
hreads.h
|
| f10ffec18f930d92e1abe9200d60e746831841a7 |
21-Nov-2013 |
Glenn Kasten <gkasten@google.com> |
Code simplification and movement to prepare for mActiveTracks
Share code to acquire wake lock at entry to loop
Change-Id: I01fb645de9417055c0adfc6a4c4011bf22ea4f94
hreads.cpp
|
| e75da4004b2c814987aa2adf8a76190f92d99c65 |
20-Nov-2013 |
Glenn Kasten <gkasten@google.com> |
Fix bug in creation of EffectHandle when out of memory
If there is insufficient client memory to create the EffectHandle,
it was returning a handle which would be useless. Now it correctly
reports the out-of-memory error back to client.
Change-Id: I894e65d5d17df39383032c1527be6ccd35f578bb
ffects.cpp
ffects.h
hreads.cpp
|
| 663c2247b71086e30bfd3192979d1dd7f15c539e |
24-Sep-2013 |
Glenn Kasten <gkasten@google.com> |
Consistent error checking for sp<IMemory> and pointer()
There have been concerns that an sp<IMemory> could be non-0, but the
associated pointer() still be NULL. There are rumors this may happen
when a non-0 sp<IMemory> is passed in by client but the shared memory cannot
be re-mapped into mediaserver.
There's also evidence in the early (2009/03/03) pre-git code of checking
pointer() for NULL, after a local allocate() returned a non-0 sp<IMemory>.
It's not clear if this is "cargo cult" paranoia, or if there was a
genuine reason for the check.
In any case, we now consistently check pointer() for sp<IMemory>
input parameters in createTrack() and queueTimedBuffer().
We also check after successful allocate(). If allocate() returns a
non-0 sp<> but NULL pointer(), then treat it as if the allocate() had
returned 0.
Change-Id: I3013ac5766b493d443ecef71711ec861076a623e
udioFlinger.cpp
racks.cpp
|
| 30ff92cba19c5acd747631365db1e1084e45ab34 |
20-Nov-2013 |
Glenn Kasten <gkasten@google.com> |
Code simplification for timed allocate
Change-Id: I5acccd7418c5b4a4dc0112147c750fef9f74bf33
racks.cpp
|
| 731076cc7857fe322eed018569a2a67f6eac0e54 |
15-Nov-2013 |
Eric Laurent <elaurent@google.com> |
am 06d62548: am d7e59228: audioflinger: do not use raw pointer for tracks
* commit '06d6254823b8f7a46690369e57b5f149c94c9f65':
audioflinger: do not use raw pointer for tracks
|
| d7e59228caad3867794d847f6bf163c6495e9506 |
15-Nov-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: do not use raw pointer for tracks
Commit 9da3d95 surfaced a problem caused by the use of a raw
pointer to a track in offload thread implementation.
Pointers to tracks should always be weak or strong pointers.
Bug: 11708529.
Change-Id: Ic48632532d186c9be8261f73cefdf824b9fbbd2b
hreads.cpp
hreads.h
|
| 561879df42b65f44b83c391d78d351e9ad4cb162 |
14-Nov-2013 |
Eric Laurent <elaurent@google.com> |
am 307aa870: am 4215e616: Merge "audioflinger: fix offload track transition" into klp-dev
* commit '307aa87038017dff8ae1dd0262c0ff87723894d7':
audioflinger: fix offload track transition
|
| b04b25321dd323142eec7c5136c81dea3629cf14 |
14-Nov-2013 |
Eric Laurent <elaurent@google.com> |
am b50fa77b: am 9f357f31: Merge "audioflinger: fix offload resume after drain" into klp-dev
* commit 'b50fa77b2cc7dadae1b8ebb591d42976879ac43c':
audioflinger: fix offload resume after drain
|
| 9da3d9573a18ffe08365557c706cf52f09118d1c |
13-Nov-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix offload track transition
Make sure that when transitioning from one active
offloaded track to the next we flush DSP content
if both tracks are not on the same audio session.
This happens when switching between two applications
playing the same type of content (e.g MP3, stereo, 44.1).
In this case, we reuse the same output thread because the track
formats are compatible.
Bug: 11247103.
Change-Id: I2b9031591149adeb70766da5e0d21ff2933a37e8
hreads.cpp
|
| 1b9f9b134e732a48198e51f16424f330cbf03143 |
13-Nov-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix offload resume after drain
When pausing and resuming during the drain,
the drain sequence number shuold not be modified
otherwise the drain callback will be ignored.
This causes failure to notify end of stream to audio
player and transition to next song.
Bug: 11247103.
Change-Id: I2a35c5cc3fd6aa667cdd257f9e9cc8715cef5159
hreads.cpp
|
| 579dd27d96497022e534e859c6ebbec675ee07aa |
08-Nov-2013 |
Glenn Kasten <gkasten@google.com> |
Fix build
Change-Id: I1ecccada22047d9c828152a53ad68b5599ff8827
hreads.cpp
|
| 8594843c15b4722ced39436fe9e64f3e57e7ace4 |
19-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Pull HAL reads out of RecordThread resample
Previously, the resample was indirectly calling HAL read
from within the BufferProvider's getNextBuffer. Now the HAL
read is done ahead of time into a circular buffer,
and the getNextBuffer just reads out of that memory.
This change will permit some future planned features.
Also removed the mono to mono resampling optimization,
as that optimization will no longer be valid in the future.
Change-Id: I59f5016a2df91078d697c37a7dec2d9d9f44542e
udioFlinger.cpp
hreads.cpp
hreads.h
|
| d8a1de6204829b0379bbd13042f1450bdb203f2b |
08-Nov-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Channel counts are uint32_t"
|
| f551e9961a6f66d63bb57f288c4be01b00725289 |
20-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Channel counts are uint32_t
Change-Id: I5676b6a291b72d7db6326f7eaf6d17588de51b05
udioMixer.cpp
hreads.cpp
|
| 09faf14b434bec0d184fecec947e3d750bcc64ff |
07-Nov-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Rename standby methods"
|
| d6fadf0479b489b09b764070974d8a59855ede64 |
30-Oct-2013 |
Glenn Kasten <gkasten@google.com> |
Simplify track 'needs' bits
Use more standard coding convention for bit masks,
and add a FIXME about max channel count.
Change-Id: I856784016703417ee480b92ae73757c472f9cf95
udioMixer.cpp
udioMixer.h
|
| 8f32537d028231abed103c68705bc5d07cedf919 |
30-Oct-2013 |
Glenn Kasten <gkasten@google.com> |
Remove redundant this->
Change-Id: Id783201e802b6ce294fa92f4bbe492adfb446686
udioMixer.cpp
|
| e25b778b11183fda65418ab2142873a030521870 |
30-Oct-2013 |
Glenn Kasten <gkasten@google.com> |
Remove dead code
Change-Id: Ide9196aa348cd3846b5c71d51fe6538f38ea8dc2
udioMixer.cpp
|
| 5c6ba3ad8eed1d59e5e2ffe6b70ced2d582a9def |
06-Nov-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix race condition in initializing downmix"
|
| 49c34acbef91800930b0fe561e0787145a10cfcc |
30-Oct-2013 |
Glenn Kasten <gkasten@google.com> |
Fix race condition in initializing downmix
The code that initializes downmix GUID searches for an effect of
the right type. If two or more AudioMixers are created, there is a
possibility that the GUID will temporarily have the wrong value.
Now the initialization is only done once, regardless of how many
AudioMixers there are.
Other minor changes:
- use naming convention of leading 's' for static member variables
- log message for no downmix effect is now a warning instead of an error,
since it is not an error until there is actual content to downmix
without a downmix effect
Change-Id: I33ab0e51cf9d24ecaf8ae868883736b0772178c4
udioMixer.cpp
udioMixer.h
|
| 0699ed950b307663e0f7e01aecdfb505e62df7d6 |
06-Nov-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Move local declarations to point of first use"
|
| 44f412a81dcca8f7149d483eeeee9057ad0101cf |
06-Nov-2013 |
Eric Laurent <elaurent@google.com> |
am 44dc9155: am 48b6cd98: Merge "audioflinger: fix direct output underrun" into klp-dev
* commit '44dc91554521eaab478674a7f812cb8a4fa4f809':
audioflinger: fix direct output underrun
|
| 7bb88779cdc1a1fe1445d6d1691fb7daac932e74 |
06-Nov-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Rename TrackPaused to trackPaused"
|
| 1b3ec0f33cbdcac5e148c23abc5bc4f68d780ed4 |
06-Nov-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Use const more places"
|
| a23f17ac334ff20a11ee63dd177cb1080e44c483 |
06-Nov-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix direct output underrun
Underruns on tracks handled by a direct or offloaded
threads were not properly reported to the applications
causing a failure to automatically recover.
Bug: 11535001.
Change-Id: I7a8696b4d646f78b16710addfe7c108d0dd0038d
hreads.cpp
|
| 93e471f620454f7de73d190521568b1e25879767 |
19-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Rename standby methods
ThreadBase::standby() returns the value of mStandby.
RecordThread::standby() "overrode" ThreadBase::standby(), but with a
different meaning and no return value. This meant you couldn't
access the parent's class definition within RecordThread.
This could be confusing, so this CL renames RecordThread::standby()
to standbyIfNotAlreadyInStandby().
Change-Id: I9196ee12701dd2dccdfaed922af2d7b8e8774676
hreads.cpp
hreads.h
|
| b91aa63779ad1ce1846fbc604f9ec935e0c67852 |
19-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Move local declarations to point of first use
Change-Id: I8a078386e371caab5174fc07b47f48fb9f21609a
hreads.cpp
|
| e6f35b13567d920af19f0b8dc35f9120b7cf1dc9 |
19-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Rename TrackPaused to trackPaused
Change-Id: I3b38e0c9a6bbb6e18cb5b1f72670139a3071cb8f
hreads.cpp
|
| b2737d0b33c17e408d96d6f9eeaa3381479c94c7 |
19-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Use const more places
Change-Id: Ibc068d319d6fff26f2d11248e17481d8f7f027e0
udioFlinger.h
hreads.cpp
|
| 1c7258b3c34b5656de92d845f6366a0f3ac1e0c6 |
31-Oct-2013 |
Eric Laurent <elaurent@google.com> |
am 4cf799fb: am 5c7ffaed: Merge "AudioFlinger: fix duplicating thread standby." into klp-dev
* commit '4cf799fbbb74de102d34ccb014ba7fbd9bfa5e14':
AudioFlinger: fix duplicating thread standby.
|
| 5c7ffaed6f04b844c2e4e2024bfb0203ac2233dd |
31-Oct-2013 |
Eric Laurent <elaurent@google.com> |
Merge "AudioFlinger: fix duplicating thread standby." into klp-dev
|
| 2c3740f01acca69c3e0bcc5e01bb0edc51b6777f |
31-Oct-2013 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: fix duplicating thread standby.
Commit fd47797 fixing issue 11247103 introduced a regression
where the duplicating thread does not exit standby state as
expected. This in turn prevents the tracks sending audio to
other threads to complete their activity cycle normally.
The fix conists in clearing mStandby state also in the specialized
threadLoop_write() method of DuplicatingThread subclass.
Bug: 11451415.
Change-Id: Ie8a92c819a56a2834ea25fa70a8b7b2125721775
hreads.cpp
|
| 1fc6cb5be26734938b3999e6bf4ce0e625e7d284 |
30-Oct-2013 |
Eric Laurent <elaurent@google.com> |
am ce275b7d: am 1c594b63: Merge "audioflinger: enable effects after registration if needed" into klp-dev
* commit 'ce275b7dd20e3d4781a46126c753e6eb998e6c0f':
audioflinger: enable effects after registration if needed
|
| 1c594b637df26499ce1dae2db34f2b3290efd838 |
30-Oct-2013 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: enable effects after registration if needed" into klp-dev
|
| d21952fe5bc94f7bb6ee729f806c0e641fdb1a0d |
29-Oct-2013 |
Eric Laurent <elaurent@google.com> |
am 7f8f7585: am fd477974: AudioFlinger: more fixes for offload audio lost
* commit '7f8f758506a95d6fe5c0e52aa7b5ef82fb18861a':
AudioFlinger: more fixes for offload audio lost
|
| fd4779740ec3e9e865d5514464df26d015354388 |
26-Oct-2013 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: more fixes for offload audio lost
mStandby being false was not a reliable indication that
something had been written to audio HAL.
Considering the last track in mActiveTracks vector as the
latest activated track was wrong as this is a SortedVector.
Bug: 11247103.
Change-Id: I397d011c5afcdd779def3fe25aaae7669a472cfc
hreads.cpp
hreads.h
|
| 9cae217050aa1347d4ac5053c305754879e3f97f |
14-Jan-2013 |
Marco Nelissen <marcone@google.com> |
Assign blame for playback wakelocks.
Set a work source for the playback wakelock, so that playback is
counted against the requesting app instead of the media server.
Cherrypicked from master.
b/9464621
Change-Id: I7329f88a288a95a582a78005a1c3d16a5a611e31
udioFlinger.cpp
udioFlinger.h
laybackTracks.h
ecordTracks.h
hreads.cpp
hreads.h
rackBase.h
racks.cpp
|
| 462fd2fa9eef642b0574aa7409de0bde3fec8d43 |
14-Jan-2013 |
Marco Nelissen <marcone@google.com> |
Assign blame for playback wakelocks.
Set a work source for the playback wakelock, so that playback is
counted against the requesting app instead of the media server.
Change-Id: I7329f88a288a95a582a78005a1c3d16a5a611e31
udioFlinger.cpp
udioFlinger.h
laybackTracks.h
ecordTracks.h
hreads.cpp
hreads.h
rackBase.h
racks.cpp
|
| a153b68f2260a8ed7fbb236fa659b13264ac5ac0 |
19-Oct-2013 |
Glenn Kasten <gkasten@google.com> |
am 77177fa2: am 95c03858: Merge "Allow releaseBuffer after flush" into klp-dev
* commit '77177fa20773d02b4f9c4147ecb98107f019fa7d':
Allow releaseBuffer after flush
|
| 95c03858e2ab4fb693a2bfe47b3caa806e43c044 |
19-Oct-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Allow releaseBuffer after flush" into klp-dev
|
| 2e422c472c91aa7912befd0fc038d1e11f354bc1 |
18-Oct-2013 |
Glenn Kasten <gkasten@google.com> |
Allow releaseBuffer after flush
After AudioTrack start checks for pending flush,
allow releaseBuffer on any previously obtained buffer.
For example, this can happen if the resampler has obtained
a buffer but not released the whole buffer yet.
Note that the resampler will be reading obsolete data.
Bug: 11285590
Change-Id: I0614fbb62e43604aac3089cce4b7797c87a306b5
racks.cpp
|
| 6a7327f110b86613e6e8c7de105c49803acacc1d |
18-Oct-2013 |
Eric Laurent <elaurent@google.com> |
am 6edd8ca5: am 6a51d7ed: audioflinger: fix track terminated before playing
* commit '6edd8ca5b7e3d169db4314d2b994bc9d48e9e082':
audioflinger: fix track terminated before playing
|
| 6a51d7ed7062536ccc892c8850a34ed55cbc8d5c |
18-Oct-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix track terminated before playing
When a track is terminated before any audio was written,
we should not send a drain request to the HAL. The drain
will be ignored and we will wait the async callback for ever.
Also fix a potential deadlock by not deleting a track with the
thread mutex held: mPreviousTrack changed from a sp to a raw pointer.
Bug: 11247103.
Change-Id: I0a4b47b099204c7c326d45d700234ab8f72d8eae
hreads.cpp
hreads.h
|
| 9138d32050411c04cc45c25094352cf8b6e01df9 |
18-Oct-2013 |
Glenn Kasten <gkasten@google.com> |
am e45796de: am dedc7b0f: Merge "Fix flush() followed by start()" into klp-dev
* commit 'e45796dee9c072cdb13a00b407e68be2cd55e2f2':
Fix flush() followed by start()
|
| 12022ff8d223ccf4dfae019c81e4a93a3f44985c |
17-Oct-2013 |
Glenn Kasten <gkasten@google.com> |
Fix flush() followed by start()
Bug: 11259166
Change-Id: I72526cdfd4e3ff4b06e6a317e577f57cc285205d
racks.cpp
|
| 3278eed31fcf82045ac00dcd549dda5270c88971 |
15-Oct-2013 |
Narayan Kamath <narayan@google.com> |
am 8580679e: am 014e7fa2: Make Audio wakelock names less unique.
* commit '8580679e1780617d978d28a43e932314b86608ce':
Make Audio wakelock names less unique.
|
| e6056ba0f2399fa3ebea9665e12a8237f99de250 |
15-Oct-2013 |
Yu Yezhong <yuyezhong@gmail.com> |
Frameworks: AudioFlinger: Fix effects memory leak
Patch by: csheng@marvell.com
Change-Id: I4e3032fb34643902f98e8fd7c8e894b6452324c9
Signed-off-by: Yu Yezhong <yuyezhong@gmail.com>
udioPolicyService.cpp
|
| 014e7fa2e90827d911c37bb0ce4d2e10e14d0bb3 |
14-Oct-2013 |
Narayan Kamath <narayan@google.com> |
Make Audio wakelock names less unique.
These wakelocks now show up in the client process, where they
might cause wakelock overflows if AudioFlinger threads die often.
The client process should be agnostic of audio flinger threading,
so don't make the wakelock names rely on audioflinger thread names.
Wakelock names now depend only on the audio flinger thread type, and
not the ID itself.
bug: 11200551
Change-Id: Ia1f71b569cb7502d617b77596abc245f461a6ec0
hreads.cpp
hreads.h
|
| d72b7c0180ee83fc3754629ed68fc5887a125c4c |
13-Oct-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: enable effects after registration if needed
This fixes a bug where effects were not properly reenabled after
being moved from one playback thread to another. The effect
is enabled but the audio policy manager sees it as disabled.
Bug: 11181933.
Change-Id: I19cac7acbaf61c546e667fd85ed7d4eda1c716d8
udioFlinger.cpp
racks.cpp
|
| 35295078ab59c8c5d143a54d5a55557c3ca62c51 |
07-Oct-2013 |
Glenn Kasten <gkasten@google.com> |
resolved conflicts for merge of 9a7d11f0 to master
Change-Id: Ie4d101185a93d37ec280f608813ad62a077a584e
|
| fee4ce338d78eeb58af1f66831ead53322d3859e |
07-Oct-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Cleanup openRecord error handling" into klp-dev
|
| 61f58c0c8d02970ea6d94ff816c54bf606f755b7 |
05-Oct-2013 |
Eric Laurent <elaurent@google.com> |
am c22a1a49: am ea0fadeb: audioflinger: offload: fix pause/flush/resume
* commit 'c22a1a495c26b8a36fa15233cc136cb6ce6d6bea':
audioflinger: offload: fix pause/flush/resume
|
| ea0fadeb5d81ef3cb7f9db458c9033d628bdb86a |
05-Oct-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: offload: fix pause/flush/resume
If a pause/flush/resume sequence is fast enough, resume is received while
we are still in PAUSING state in which case it is a NOP. If this happens,
flush is still forwarded to the audio HAL but is not preceeded by a pause
which can cause problems to the audio DSP.
It is necessary to preserve the flush as this sequence is typical to a seek.
The fix consists in forcing a pause/resume when a flush request must be
executed and the audio HAL has not been paused previously.
Bug: 11081559.
Change-Id: Ib84ed26d503a61c05933b923ec556b10cedfe140
hreads.cpp
|
| 1c7feca9cc3e408dfeb632e7060cf31e5015227e |
04-Oct-2013 |
Eric Laurent <elaurent@google.com> |
am 6c48a011: am 026c5b07: Merge "audioflinger: fix wake lock acquisition check" into klp-dev
* commit '6c48a01188746202164f8e10d63ff3775dfcf727':
audioflinger: fix wake lock acquisition check
|
| 547789d25dc6bd6561553bcf6b384fb0d4fee834 |
04-Oct-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix wake lock acquisition check
Commit e14a5d6 introduced a bug in ThreadBase::acquireWakeLock() where
the wake lock acquisition return code is not initialized before
being checked causing the wake lock token to be sometines not set
and potential failure to release the wake lock later.
Bug: 10985160.
Change-Id: Iffd40e773ae3bcfec3c148a844d5dbebbf474eaf
hreads.cpp
|
| 5918e0cb64200f64632595f7aea6062c65d43353 |
04-Oct-2013 |
Marco Nelissen <marcone@google.com> |
am b5677882: am 8097068d: Merge "Better recording wake lock accounting" into klp-dev
* commit 'b5677882cf9a34bd3f2daf76d10951282ea9b083':
Better recording wake lock accounting
|
| 8097068de97c0cdab8150931f290edbd5b3721e5 |
03-Oct-2013 |
Marco Nelissen <marcone@google.com> |
Merge "Better recording wake lock accounting" into klp-dev
|
| e14a5d6d2cc91dd2fc09ffdf7aa670b37da0795d |
03-Oct-2013 |
Marco Nelissen <marcone@google.com> |
Better recording wake lock accounting
Count wake lock use against the app that initiated the recording,
not against the media server.
b/10985160
Change-Id: Iae6e6c030b7f1c6081d9c79725d6990f60dadaaa
hreads.cpp
hreads.h
|
| 34717c83733def81287e2b4ba2f62b416325c7ae |
02-Oct-2013 |
Eric Laurent <elaurent@google.com> |
am 3424d6e1: am 1adf20ce: Merge "fix volume and effect enable delay on offloaded tracks" into klp-dev
* commit '3424d6e17637e0743ddf3bf4688af8ee36e69264':
fix volume and effect enable delay on offloaded tracks
|
| 98232ab7c20bfbdfe176d2c25bb810dbb5e7b4f1 |
30-Sep-2013 |
Jean-Michel Trivi <jmtrivi@google.com> |
am a2340b30: Merge "Don\'t call audio effect process on offloaded playback threads" into klp-dev
* commit 'a2340b3059b9d6e4286d0db7069df477c164df0c':
Don't call audio effect process on offloaded playback threads
|
| a6c7d98033c161592a062c61eaa6b7fd635e3006 |
30-Sep-2013 |
Eric Laurent <elaurent@google.com> |
am 0adc67df: Merge "audioflinger: fix crash when starting offload thread" into klp-dev
* commit '0adc67dfc5cedb211c36f06849681a60a32f5805':
audioflinger: fix crash when starting offload thread
|
| 59fe010bcc072597852454a2ec53d7b0a2002a3b |
28-Sep-2013 |
Eric Laurent <elaurent@google.com> |
fix volume and effect enable delay on offloaded tracks
Volume: add a method to wake up the mediaserver playback
thread when a volume command is received on an offloaded track.
Effects: call effect chain process on offloaded playback threads
asynchronously from writes to allow effect state updates while
waiting for async write callback.
Bug: 10796540.
Change-Id: Id2747ae88783575d1d7ffd6fc86fbd054ab2c739
udioFlinger.h
ffects.cpp
laybackTracks.h
hreads.cpp
racks.cpp
|
| a2340b3059b9d6e4286d0db7069df477c164df0c |
27-Sep-2013 |
Jean-Michel Trivi <jmtrivi@google.com> |
Merge "Don't call audio effect process on offloaded playback threads" into klp-dev
|
| 4de95592980dba88a35b3dc8f3fd045588387a4f |
27-Sep-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix crash when starting offload thread
Passing a sp to parent thread to AsyncCallbackThread() constructor
causes a strong reference to be acquired on the OffloadThread inside
its constructor which causes an early launch of the thread loop
with unpredictable consequences.
Pass a wp to parent thread instead.
Also move the creation of the AsyncCallbackThread to
readOutputParameters() where mUseAsyncWrite is initialized which
makes more sense.
Also change the type of AsyncCallbackThread parent thread to PlaybackThread
instead of OffloadThread to allow a broder use of non blocking write which
in theory is not limited to audio offload use case.
Bug: 8174034.
Change-Id: I4b093b022030cd4f5eb8b8e477333e91098a6549
hreads.cpp
hreads.h
|
| fed6292af65a0b97b583ecbd3c232b3811a3f37b |
26-Sep-2013 |
Jean-Michel Trivi <jmtrivi@google.com> |
Don't call audio effect process on offloaded playback threads
An audio effect process command is not meant to be called for
playback threads belonging to offloaded playback.
Bug 10933817
Change-Id: Idc7b6a0224791bbf8d61648474d3c288617c498f
ffects.cpp
|
| 62dbe93f7743061e97fc3299386a2f887ca748cf |
25-Sep-2013 |
Eric Laurent <elaurent@google.com> |
am 4e141a4f: am 1802399c: Merge "fix deadlock in audioflinger::createEffect()" into klp-dev
* commit '4e141a4f1303767f70ea983925ec01b034fe667b':
fix deadlock in audioflinger::createEffect()
|
| eb3c337a3d6c74ec857dfc8be7eeafe634614bcd |
25-Sep-2013 |
Eric Laurent <elaurent@google.com> |
fix deadlock in audioflinger::createEffect()
commit 5baf2af5 introduced a regression by calling
getOutputForEffect() with AudioFLinger main mutex locked.
The locking order must always be AudioPolicyService mutex then
AudioFlinger then ThreadBase mutex.
Bug: 10916796.
Change-Id: Ide34a2d84dbb06dbb35abd0640d91b01b0ac4d40
udioFlinger.cpp
ffects.h
|
| e93cf2ca27ae6f4a81d4ef548bbf10a34db6d98f |
24-Sep-2013 |
Glenn Kasten <gkasten@google.com> |
Cleanup openRecord error handling
Bug: 10888816
Change-Id: I84897dd7d30b370640b54e928f230604b873cb68
udioFlinger.cpp
hreads.cpp
|
| c0dab528e8c63abce2022a391dface95c3ef67e5 |
24-Sep-2013 |
Eric Laurent <elaurent@google.com> |
am a57e2ab5: am 664539d2: audioflinger: fix ro.audio.silent in offload again
* commit 'a57e2ab58a17d3da35593a6c85f65e19249e9ebc':
audioflinger: fix ro.audio.silent in offload again
|
| 79360ad49d6f84d9e99a08d225a642d65079d740 |
24-Sep-2013 |
Eric Laurent <elaurent@google.com> |
am 3f1d3b97: am 21e54568: fix incoming call audio start time.
* commit '3f1d3b97f723cf39fe6bde9efd0a3fc9e78028a5':
fix incoming call audio start time.
|
| 664539d25180ab8f77e0521533ea2821cf28985f |
24-Sep-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix ro.audio.silent in offload again
Commit 1abbdb4 was not working 100% of the times because
it is possible that the offload thread loop never sleeps
after being created in which case the property is never read.
The loop now reads the property once when starting.
Bug: 10899309.
Change-Id: I2e2ca332f2d7086e59d65f6010378c4d9618ba9e
hreads.cpp
|
| 21e5456821e3c107d09b0629adbf23c9e06e7c0b |
23-Sep-2013 |
Eric Laurent <elaurent@google.com> |
fix incoming call audio start time.
commit cec4abb fixing issue 6720482 had a flaw that
would insert commmands at the wrong position in the
audio policy service command thread queue causing other
commands to be delayed more than expected.
A change in track stop timing introduced by commit bfb1b83
made this issue visible when answering a call.
Bug: 10860012.
Change-Id: Iec769112f14e13d13b64628b808e43663e33771c
udioPolicyService.cpp
|
| 0230a2a8a413076a138db4c4e1dea018104fd5e2 |
20-Sep-2013 |
Eric Laurent <elaurent@google.com> |
am c6bed216: am a3b97ad2: Merge "audioflinger: implement getTimestamp() for offloaded tracks" into klp-dev
* commit 'c6bed216e3a759a855ac4b9b743fbaed2d5929c1':
audioflinger: implement getTimestamp() for offloaded tracks
|
| d2b80a1fb90cb4dc3f569e716af0279c1e1ea72d |
20-Sep-2013 |
Eric Laurent <elaurent@google.com> |
am ce3e7cb0: am fbb2609f: Merge "audioflinger: fix lost offload thread resume event" into klp-dev
* commit 'ce3e7cb0bd09e7861d417b7d45c1e5921f10a93f':
audioflinger: fix lost offload thread resume event
|
| a3b97ad288e6808c54c6275eaac1bd53108d8f11 |
20-Sep-2013 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: implement getTimestamp() for offloaded tracks" into klp-dev
|
| accc147666bfd37fc8b4ef745f18a8c751555ec2 |
20-Sep-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: implement getTimestamp() for offloaded tracks
Bug: 9587132.
Change-Id: Ie9d5f4cca96306d08bc9a2dbd6edd8953096702d
hreads.cpp
hreads.h
racks.cpp
|
| fbb2609f132f23a983d376dec9ae6bfebfcacc4c |
20-Sep-2013 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: fix lost offload thread resume event" into klp-dev
|
| ede6c3b8b1147bc425f7b923882f559a513fe23b |
19-Sep-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix lost offload thread resume event
It was possible that a resume request signaled by addTrack_l()
while waiting for an async write callback is lost. This is because
mSignalPending was not set and waitingAsyncCallback_l() would pause the
thread loop before executing prepareTracks_l().
The fix consists in using signal_l() method to wake the thread
loop o that mSignalPending is set.
Also make sure that sleepTime is reset to 0 when resuming to make sure
that we write any remaining bytes to the HAL.
Bug: 10810347.
Change-Id: If9a3b22cc3b9e6eb384a56c48c40e6258d0896ad
hreads.cpp
hreads.h
racks.cpp
|
| d7e076589dc5298d7a78cb683159723b7eb08d7f |
19-Sep-2013 |
Eric Laurent <elaurent@google.com> |
am 87fb4f30: am d1b28d41: audioflniger: fix leaked audiosession references
* commit '87fb4f30077e7bcf78e65cf25fe95bd58238bad1':
audioflniger: fix leaked audiosession references
|
| d1b28d41dbda203ffb420ba2e36cbe736b163ff8 |
19-Sep-2013 |
Eric Laurent <elaurent@google.com> |
audioflniger: fix leaked audiosession references
acquireAudioSessionId() should not create session references
if the caller is not listed in the known notification clients.
It happens when the MediaPlayer or AudioTrack is created by the
mediaserver on behalf of a client (e.g CameraService). In this case
releaseAudioSessionId() can be called from a different pid and
the session reference will not be removed.
Also we will not be able to remove the reference if the client
process dies.
Bug: 10606426.
Change-Id: Ibb5e27518daeb484a2e66302e4c6221cbaca5b70
udioFlinger.cpp
|
| dc8cae8c118e4aef4ef1f7b2c6f79becc1df4a05 |
18-Sep-2013 |
Eric Laurent <elaurent@google.com> |
am bf5e2397: am 5baf2af5: more support for audio effect offload
* commit 'bf5e23979a03da96ce1d63126c480103232f174b':
more support for audio effect offload
|
| 5baf2af52cd186633b7173196c1e4a4cd3435f22 |
13-Sep-2013 |
Eric Laurent <elaurent@google.com> |
more support for audio effect offload
Offloading of audio effects is now enabled for offloaded
output threads. If an effect not supporting offload is enabled,
the AudioTrack is invalidated so that it can be recreated in PCM
mode.
Fix some issues in effect proxy related to handling of effect
commands to offloaded and non offloaded effects.
Also fixed a bug on capture index in software Visualizer effect.
Bug: 8174034.
Change-Id: Ib23d3c2d5a652361b0aaec7faee09102f2b18fce
udioFlinger.cpp
udioFlinger.h
ffects.cpp
ffects.h
hreads.cpp
racks.cpp
|
| 6d07d9bd2e0549ca2682ad9e94118feee1f733f6 |
14-Sep-2013 |
Eric Laurent <elaurent@google.com> |
am 029838a9: am 7f91e5e0: Merge "audioflinger: fix ro.audio.silent in offload mode" into klp-dev
* commit '029838a98c9377299ef3341e21e9378fd5562ab6':
audioflinger: fix ro.audio.silent in offload mode
|
| d455cdfad40ca0558b8f4f800ec192027e272c14 |
14-Sep-2013 |
Eric Laurent <elaurent@google.com> |
am c91eb605: am 972a173d: audioflinger: longer offload thread standby delay
* commit 'c91eb6059a6af8b84c48e0ce727b01829f9cd789':
audioflinger: longer offload thread standby delay
|
| 7f91e5e09cf3bd8068894b2e54cbdaac70bf53a0 |
14-Sep-2013 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: fix ro.audio.silent in offload mode" into klp-dev
|
| 1abbdb4429479975718421c4fef3f79ce7c820e3 |
14-Sep-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix ro.audio.silent in offload mode
Change-Id: I5a1a79000d53146689b0a198cc5419c36509703f
hreads.cpp
|
| 972a173d7d1de1a3b5a617aae3e2abb6e05ae02d |
04-Sep-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: longer offload thread standby delay
- Increase offloaded output thread standby delay to 1 second
to allow transition between tracks with going to stanby
if reusing the same audio track (gapless)
- Make sure pause/flush/resume sequence is sent to the HAL
in the right order
- Fix format display in track dump
Bug: 8174034.
Change-Id: I43ef6f8fdbf7427e4eff6cc2d0665d7d1463ea8a
hreads.cpp
racks.cpp
|
| f523897787b64cf2217d0ab4946b564420c675fe |
12-Sep-2013 |
Glenn Kasten <gkasten@google.com> |
am 8a220740: am dc33c542: Merge "Fix AudioTrack shared memory leak" into klp-dev
* commit '8a220740cd77eb844123d3914190a94513797b40':
Fix AudioTrack shared memory leak
|
| dc33c542351449ceee95cc50a9009f495e7c72ba |
12-Sep-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix AudioTrack shared memory leak" into klp-dev
|
| db358afc965d9bfb703953ac004275428a7cda64 |
12-Sep-2013 |
You Kim <you.kim72@gmail.com> |
AudioPolicyService: malloc/delete pair
Change-Id: I75cd44ac0caccda9148faaa052c9e7a0c06d46d1
Signed-off-by: You Kim <you.kim72@gmail.com>
udioPolicyService.cpp
|
| d5577f26de1ae3a0dc6fbea9c60a07d585f894bf |
12-Sep-2013 |
Eric Laurent <elaurent@google.com> |
am 4823964d: am efc4d489: Merge "audioflinger: add HOTWORD audio source." into klp-dev
* commit '4823964d198228289ff21530bf8b01bb2ef196bb':
audioflinger: add HOTWORD audio source.
|
| efc4d4899e03fb8464017dd48539106cec2163de |
12-Sep-2013 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: add HOTWORD audio source." into klp-dev
|
| 0c72b24f91c68442eb374bd1b338c394105b8262 |
11-Sep-2013 |
Glenn Kasten <gkasten@google.com> |
Fix AudioTrack shared memory leak
Bug: 2801375
Change-Id: I50e15164fe310f69ea019dca5b49171a02bc6992
laybackTracks.h
racks.cpp
|
| 9a54bc27876acd5d8be5b1fc3dc46701fe76fbb3 |
09-Sep-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: add HOTWORD audio source.
HOTWORD is a special capture audio source that uses
the same audio tuning as VOICE_RECOGNITION but allows an
active capture to be preempted.
Bug: 10640877.
Change-Id: Iea09a11cbcdbacc8d434e5230e7559dcd1b44ac0
udioPolicyService.cpp
ecordTracks.h
erviceUtilities.cpp
erviceUtilities.h
hreads.cpp
racks.cpp
|
| 8136cfae9c22ae8ff42eec9ed751833dda605444 |
09-Sep-2013 |
Eric Laurent <elaurent@google.com> |
am 8a910716: am 6ca83fad: Merge "audioflinger: no effects on offloaded tracks" into klp-dev
* commit '8a910716892d17a2ac62c7e9884af0e9d75b26bc':
audioflinger: no effects on offloaded tracks
|
| 813e2a74853bde19e37d878c596a044b3f299efc |
31-Aug-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: no effects on offloaded tracks
Invalidate offloaded tracks when an effect is enabled
so that the track is recreated in PCM mode and the effect
can be applied.
This is temporary until effect offloading is implemented.
Bug: 8174034.
Change-Id: I77b8b54a10db6cb8334be76d863ea7e720eaad09
udioFlinger.cpp
udioFlinger.h
ffects.cpp
ffects.h
racks.cpp
|
| bfc4214079875f0dc8c349e8c475d1813c234e67 |
06-Sep-2013 |
Eric Laurent <elaurent@google.com> |
am 99b84e8b: am 3b4529e0: audioflinger: remove async write race conditions
* commit '99b84e8b013a6e0b926693067b00e82cfbe2ca4f':
audioflinger: remove async write race conditions
|
| 3b4529e03c5fc7a44c22f9091ad15a269bfca3a8 |
06-Sep-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: remove async write race conditions
Remove possible race conditions between async callback
thread and offload thread when clearing and setting the
draining and write blocked flags.
Bug: 8174034.
Change-Id: I7af10491f39dc0e7d7414862a9d8e763daa2e2b7
hreads.cpp
hreads.h
|
| 344202e13a91c9d3d109fd8f4a67f5b23efa7800 |
04-Sep-2013 |
Glenn Kasten <gkasten@google.com> |
am 55817ab9: am fe346c70: Fix miscellanous AudioTrack::getTimestamp() bugs
* commit '55817ab9c1868f1e6a59155f55c768f01509cda5':
Fix miscellanous AudioTrack::getTimestamp() bugs
|
| b9258d930d67fc314373f94ab4423974e5f87e1a |
04-Sep-2013 |
Glenn Kasten <gkasten@google.com> |
am 97992cc7: am 732845c7: FastMixer computes presentation timestamps for fast tracks
* commit '97992cc71fd0bbd4aa22e5195ee7e5cde09ac254':
FastMixer computes presentation timestamps for fast tracks
|
| 1462a2b84a578b5541683c824e0f77ebf2aa82ae |
04-Sep-2013 |
Glenn Kasten <gkasten@google.com> |
am a4a82c5f: am 6466c9e6: Add ExtendedAudioBufferProvider::framesReleased and onTimestamp
* commit 'a4a82c5f8f3eccfa10ce30a52cbb4b1c5c983282':
Add ExtendedAudioBufferProvider::framesReleased and onTimestamp
|
| fe346c707f59d763ded93bc3d27b51f0c0408258 |
30-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Fix miscellanous AudioTrack::getTimestamp() bugs
Check that get_presentation_position is non-NULL before calling.
AudioTrack::getTimestamp not implemented for fast tracks.
Fix typo in Track::getTimestamp().
Fix bugs in AudioTrack::getTimestamp after stop:
- getTimestamp while stopped is not allowed.
- stop, start, getTimestamp now returns the correct value.
Change-Id: Ie8d9dc1f28d8927634e04175a68b147ffc2ea8eb
racks.cpp
|
| a7fef85e7d419a4f5d6a3144f9ba70ceff2f122a |
04-Sep-2013 |
Glenn Kasten <gkasten@google.com> |
am f81d5fc1: am 943ba3cc: Merge "Add NBAIO_Sink::getTimestamp()" into klp-dev
* commit 'f81d5fc1d67a5252d0da1a45274460c4dd33b87b':
Add NBAIO_Sink::getTimestamp()
|
| 732845c7e8d294bb1aaa4cd9687da62b51f1f6e6 |
23-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
FastMixer computes presentation timestamps for fast tracks
and forwards them to each track's ExtendedAudioBufferProvider::onTimestamp().
Change-Id: I06fb9586bb7e20d5c0289abdc8cac6cd3fd2f6e8
astMixer.cpp
|
| 6466c9e6e6278c740aed77f695f679be9f5db478 |
23-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Add ExtendedAudioBufferProvider::framesReleased and onTimestamp
and implement them in SourceAudioBufferProvider using the associated NBAIO_Source,
and in Track using the associated AudioTrackServerProxy.
Change-Id: I60dc4adba63fc1dc452ff16caf347e4a7c8242c2
laybackTracks.h
racks.cpp
|
| 767094dd98b01baf21de2ad09c27b3c98776cf73 |
23-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Add NBAIO_Sink::getTimestamp()
with a real implementation in AudioStreamOutSink
for dummy implementation initially in MonoPipe.
Use in AudioFlinger::PlaybackThread::threadLoop_write()
to keep the input to the timestamp latch up-to-date.
Change-Id: I10ef277991b63bb43d55d6f3df75116ef32246cd
hreads.cpp
|
| 0fd582e3ce5243c3e5a429fee3330aafc69b69fa |
31-Aug-2013 |
Eric Laurent <elaurent@google.com> |
am 5d8cbf41: am 6bf9ae20: audioflinger: offloaded tracks flush/pause order
* commit '5d8cbf41ee7e511d20a649e240d6aecbbac3fae7':
audioflinger: offloaded tracks flush/pause order
|
| 6bf9ae20b3bd2dbb8f2e89ee167a6785222301cf |
31-Aug-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: offloaded tracks flush/pause order
Make sure that a flush received after a pause is forwarded
to the HAL after the pause.
Change-Id: Ib3221f70c59f8b3dbbac20b23104f6b779b75be0
hreads.cpp
|
| f557872b594d15599bdb65efd91eb5adeb1d0840 |
30-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
am 908d3c09: am 51ec03c2: Merge "Implement Track::getTimestamp()" into klp-dev
* commit '908d3c09ca7f2ccb280aa5dc8d876099ff9a9d0f':
Implement Track::getTimestamp()
|
| 30c296c153447f5c2bb5251928bf4d3c4e28e552 |
30-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
am 3ae0a58a: am 9841f636: Merge "Add AudioFlinger::PlaybackThread::Track::getTimestamp()" into klp-dev
* commit '3ae0a58a83d092d96672295c8a62908478f28a9b':
Add AudioFlinger::PlaybackThread::Track::getTimestamp()
|
| ec9ad1b0947f5d6b465281312dbe92f096a8f881 |
30-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
am f94b2946: am 56b59224: Merge "Add IAudioTrack::getTimestamp()" into klp-dev
* commit 'f94b2946a511c5cbb6b9001449ca8278cb332bda':
Add IAudioTrack::getTimestamp()
|
| bd096fd9d8e5fc0e62f98807f4818a06f70d0812 |
23-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Implement Track::getTimestamp()
using a new timestamp latch in PlaybackThread, and
AudioTrackServerProxy::framesReleased() which returns mServer.
Change-Id: I1ebfba968c773faaab95648c272fd3ebd74718d6
hreads.cpp
hreads.h
racks.cpp
|
| 573d80a8f463f648a515fc0975bf83951b272993 |
26-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Add AudioFlinger::PlaybackThread::Track::getTimestamp()
with a dummy implementation initially,
and use it in AudioFlinger::TrackHandle::getTimestamp()
Change-Id: I2da88fc52a135a7f0d9fd9538986e919dc8ccd3b
laybackTracks.h
racks.cpp
|
| 53cec22821072719ee02c856e9ac2dda2496c570 |
29-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Add IAudioTrack::getTimestamp()
with dummy implementation in AudioFlinger::TrackHandle, and
implement AudioTrack::getTimestamp() using IAudioTrack.
Also document invariant that mAudioTrack and control block are always
non-0 after successful initialization.
Change-Id: I9861d1454cff7decf795d5d5898ac7999a9f3b7e
udioFlinger.h
racks.cpp
|
| 61bd7a0281ffecb99b27bff63334c559df7357e6 |
20-Aug-2013 |
Jeff Brown <jeffbrown@google.com> |
am 0541b3bb: am c8b226f4: Merge "Enforce permission for recording from remote submix." into klp-dev
* commit '0541b3bb40fa23e97e0a4599463b6afc0e271939':
Enforce permission for recording from remote submix.
|
| 893a5642871114fca3b2a00c6ff8e5699ce3e3ed |
17-Aug-2013 |
Jeff Brown <jeffbrown@google.com> |
Enforce permission for recording from remote submix.
Bug: 10265163
Change-Id: I2395036bac6e10baeaf6ea4bc3093bd9dd98742d
udioFlinger.cpp
erviceUtilities.cpp
erviceUtilities.h
|
| 7165268ffa6c7b6b405b6afad82e2a346500e8ee |
15-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Don't lock and then immediately unlock effect chains
Change-Id: Id9a4b1e9c49cecfd3a008a7814b5c14dad9a5831
hreads.cpp
|
| b86432b1d62e7c9034223f0120b9bceb44215c41 |
15-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Fix races in reading activeTrack->mState without lock
Change-Id: I8cfbd9884fa5959a846ddb52cdc797266b896769
hreads.cpp
|
| 9e98235a9e9bb870e1c76911e3b4d00386a52c39 |
14-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Simplify flow by having all mActiveTrack.clear() continue
Change-Id: Icd160b0aa79a747bd19fabaa7feee813afb155e0
hreads.cpp
|
| 5edadd46c76c5ff1c3edabf2ea943c2278e82e1c |
15-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Don't sleep with effect chains locked
Change-Id: I6eae037304d3dc8a8f3ef55c5b210fd4f8856e99
hreads.cpp
|
| ad5bcc29b216412dbc4d936c270d30620c68f361 |
14-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Copy mActiveTrack for stability and easier access
Change-Id: If0d88610ec35128054a22f42b2d79f07c4b724eb
hreads.cpp
|
| 1ba19cd7fcdf18ab6efab2a1b831affab9a46157 |
14-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Simplify code by pulling up mActiveTrack == 0 test
Change-Id: I225bb032b2fde97d891191f3599d9850c2d27bb2
hreads.cpp
|
| d9fc34fb0fcfcc739f868b116edf50c62af19d5e |
14-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Remove dead 'if'
Change-Id: I23c082f0115ae598cad293be0ac66a130afb692a
hreads.cpp
|
| 4ef0b463a56c19bad9197aa9f90d792090461429 |
14-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Fix race in checking exitPending() in RecordThread::threadLoop()
Change-Id: I97a1c5ab17a6c334d275be794392dcf6c9a88d0e
hreads.cpp
|
| 2cfbf88b89854f30b295e8ae26a031edb8d712f8 |
14-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Simplify code that checks for record standby
Take advantage of the new processConfigEvents_l()
and it's post condition that mConfigEvents is empty.
Change-Id: Icd40fb175f3e374203c9c8d2e9df2c3929bed89b
hreads.cpp
|
| 26a4029c95620a2b98187cf003cd3c58eea03747 |
14-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
checkForNewParameters_l returns a value which is currently ignored
Change-Id: I19b9cc9df094d67560f6e832d94fa284dca18687
hreads.cpp
|
| d5418eb594435c958d6c37fa9938161a0112adbd |
14-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Use Autolock where possible
Change-Id: Id0e1836831c45beda4e5cb11e983005ea699fc2a
hreads.cpp
|
| f777331418a86cd9fd709af898ef24a69967aeb4 |
14-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Add ThreadBase::processConfigEvents_l()
Change-Id: I6bcdc212a984c52ab98b0b0744943b4ecc1e1542
hreads.cpp
hreads.h
|
| c527a7c2b1bfd26e8f3086e1b653d56e521379d9 |
14-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Pull in scope of effectChains
Change-Id: Id5336f8663099088a0958fb8bd59fc5df9f92829
hreads.cpp
|
| 47c2070b2ce8aedb7300c0aad91caccf3c383841 |
14-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Add record thread locking comments and FIXMEs
Change-Id: Ia5bdc9b8b013c2e40af17c82833051290bf4df70
hreads.cpp
|
| 2d94426cd3302cb1215c92c5f1c4b90c24ceb72b |
13-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Convert if/else-if in threadLoop to switch
Change-Id: I7428abb332ead6c2a1df5903f6c4e166be1f7abb
hreads.cpp
|
| 3468e8a4d79cc6a7bb0f03f8382426195bed44df |
14-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Fix switch indentation to follow our audio style
Change-Id: I137763917962b276838715319b163c763cb54c71
hreads.cpp
|
| 34fca34606b448e6b71c2942f63cb13a0aebd620 |
13-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Explicit comparison to 0 for integral types
Change-Id: I2bfe864d1bda719581e297a38d96abfc0e3748b9
udioMixer.cpp
hreads.cpp
|
| 3255095af6677caeeebf81c8891ce8db745617ff |
06-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Move local variable declarations to point of first use
Change-Id: Ia0fc4b099a3cf8bb5b67b287e7687e9dc9691e0e
udioFlinger.cpp
|
| 6e2ebe97f2ad0a21907f20f9ee644c4eacbb7a40 |
13-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Use curly braces in 'if' to make it easier to add logs
Change-Id: I58b33fefdd8bf703647414157a99a3223be3531c
udioFlinger.cpp
udioMixer.cpp
udioPolicyService.cpp
udioResampler.cpp
udioResamplerCubic.cpp
ffects.cpp
astMixer.cpp
hreads.cpp
racks.cpp
|
| e198c360d5e75a9b2097844c495c10902e7e8500 |
13-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Whitespace
Change-Id: Id3ccc183a03421330d0498faaa62a45915cdc3d6
udioFlinger.cpp
hreads.cpp
hreads.h
|
| 2fc14730e4697a6f456b4631549c9981f6b0b115 |
05-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Comments
Change-Id: I56f82904cb3855bad6f33ad1ff4ddf66a61c5720
udioFlinger.cpp
hreads.cpp
|
| c0029a8551d65929cbf51a55bd3bae3342a1f2fe |
13-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Check input HAL channel mask instead of channel count"
|
| 030033342a6ea17003e6af38a56c7edc6d2ead01 |
07-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Clean up error handling in createTrack and openRecord
Outside callers now use initCheck() to determine whether a TrackBase
has been created successfully, instead of relying on internal knowledge.
Previously, callers needed to know that a TrackBase was only valid if it's
getCbk() != 0. For a Track (playback), they needed to know to also check
the track's name (track index). Now, outsiders can just call initCheck().
Other changes:
- Return a 0 reference if track creation fails
- Remove a dead line of code in AudioFlinger::openRecord
Change-Id: If374924a3f6fd27906f625aa83dd0a1e3f506e00
udioFlinger.cpp
laybackTracks.h
hreads.cpp
rackBase.h
racks.cpp
|
| 9156ef3e11b68cc4b6d3cea77f1f63673855a6d1 |
07-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Status pointer passed by caller is always non-NULL
in createTrack, openRecord, createEffect, createTrack_l,
createRecordTrack_l, and createEffect_l.
Change-Id: I2e459e4de9c78145f4d496e6abf289479a2f0941
udioFlinger.cpp
udioFlinger.h
hreads.cpp
hreads.h
|
| e51a70ab6326715f614c737de1e8ea1941057b3b |
12-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove obsolete TrackBase::getBuffer()"
|
| e92db4bdb575d897d6bbd5086761fa9387393c83 |
12-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Pull up readyToRun into ThreadBase"
|
| b0e6b967c290c905c4c696df4a8ac9e9d591928b |
12-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Simplify the code that allocates mMixBuffer by removing mAllocMixBuffer"
|
| 309f7abb3c170ba764e67e9b6fed31fb442e9953 |
09-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Use 0 for sp<>, NULL for raw pointers"
|
| 3a6d5ded7187ae324532a60b03532c33d607e282 |
09-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Rename reqChannels to reqChannelMask"
|
| f874f1375d758c8a7a84572a94371fbd4085f0df |
08-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Pull up mBufferSize from RecordThread to ThreadBase"
|
| 2c03d0a2b741310c47d5f625cbdec4a4e4a0778e |
08-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove obsolete field RecordTrack::mAudioRecordServerProxy"
|
| ec3fb50c8ee25ac263a9c2cbe9242f877bef960c |
17-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Check input HAL channel mask instead of channel count
The only supported channel masks are mono and stereo;
arbitrary channel masks are not yet allowed.
Change-Id: I9cebbe56480b53b98e59aa55245a1f03a30d2839
hreads.cpp
|
| aac30336994235e42a92861eb83289131a6b94dd |
06-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Remove obsolete TrackBase::getBuffer()
Change-Id: Ic39c7baa9538080bfec114bf18ff9b9986c93fed
rackBase.h
|
| 937098b9b564a779ff1c8c2d9e60769ee5c69810 |
26-Jun-2013 |
Glenn Kasten <gkasten@google.com> |
Use 0 for sp<>, NULL for raw pointers
Change-Id: Ia0595afdc41e1dec428d7a3db73558b54c417bf3
udioFlinger.cpp
hreads.cpp
|
| fb872cc6f77f6d74011fab703b3edd7023c736cb |
06-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
memset audio_config to zero
Per guideline in hardware/libhardware/include/hardware/audio.h,
"You should memset() the entire structure to zero before use to
ensure forward compatibility."
Change-Id: I2bce70ef6ea4b070668bd9cbc54bfe20a60111aa
udioFlinger.cpp
|
| f506e9495f2b2123b3bec7d42ec8ef13c6213ee8 |
06-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Rename reqChannels to reqChannelMask
This avoids potential ambiguity with channel count.
Change-Id: I0db82efc4c14dd8189d65a1db29809a085acf528
udioFlinger.cpp
|
| 6ae6b811666865815ebb1f670aacb1a0f2edaa73 |
06-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Remove obsolete field RecordTrack::mAudioRecordServerProxy
Change-Id: I054ba929e209797542940a0e734fc15f19f321c9
ecordTracks.h
racks.cpp
|
| cf04c2cb8e031acc03c1c91cb1ccab15098c89b6 |
06-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Pull up readyToRun into ThreadBase
Change-Id: Ib150afd72c3b35d83599a61664e89302a8feb4b5
hreads.cpp
hreads.h
|
| c1fac191069774c7bfcb062edbb821ea56e7dbc0 |
06-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Simplify the code that allocates mMixBuffer by removing mAllocMixBuffer
Change-Id: I5ac41210251789a034d760309c50fe018f2442ce
hreads.cpp
hreads.h
|
| 70949c47fbae3f836d15f040551d7631be3ed7c2 |
06-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Pull up mBufferSize from RecordThread to ThreadBase
Now both playback and record threads have a HAL buffer size for fast
tracks, and a potentially larger-sized buffer for normal tracks.
Change-Id: I6f57b85ecece3cbd4e9e64b8c41c3352f7656912
hreads.cpp
hreads.h
|
| 90e58b1fefc8caf70b34301a92bc86179580b6fc |
01-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
RecordThread::createRecordTrack_l use flags for fast tracks
Look at client's request for a fast track in the flags parameter,
and check whether the request can be satisfied.
Change-Id: I65ddaeb1b85670fe9066076f638bfed0cda01c0d
hreads.cpp
hreads.h
|
| ddb0ccf3fb6fe8da8c71a6deb30561b821f3c0a2 |
01-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
RecordTrack::createRecordTrack_l flags are in/out
The flags are currently unused, but will be used for requesting a fast
track. Making flags in/out will allow reporting back up to client that
the request is denied.
Change-Id: Ifbee57da3632ce130551065a426577fb97b1a68d
udioFlinger.cpp
hreads.cpp
hreads.h
|
| eeca32671896739e84050da5992d5f151a1629de |
01-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
IAudioFlinger::openRecord track_flags_t flags is in/out
This will allow AudioFlinger to tell client it is denying a request.
Change-Id: Iff2be3ad6636371bbda9c9899a283c94620c1f06
udioFlinger.cpp
udioFlinger.h
|
| f36dbfe3d517e9311be0a4d0fd6658380690e84c |
02-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Report underruns for fast tracks also"
|
| 1a3739169becc850999c3c06db402541119a428c |
01-Aug-2013 |
Glenn Kasten <gkasten@android.com> |
am 6d4d8790: am b687ff9f: Merge "Fix mismatched delete/new couple"
* commit '6d4d879037a495f9a4276a6996d1845594ce9090':
Fix mismatched delete/new couple
|
| 82aaf94a5b18939e4d790bbc752031f3070704a3 |
18-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Report underruns for fast tracks also
This fixes a regression that was introduced earlier
by commit 9f80dd223d83d9bb9077fb6baee056cee4eaf7e5
called "New control block for AudioTrack and AudioRecord".
That commit broke underrun reporting for fast tracks.
Also remove Track::mUnderrunCount, which counted the number of underrun
events, and was only used by dumpsys media.audio_flinger.
Now dumpsys media.audio_flinger reports the number of underrun frames,
Isolated underrun-related control block accesses via the proxy, so that
the server is not directly poking around in the control block.
The new proxy APIs are AudioTrackServerProxy::getUnderrunFrames() and
AudioTrackServerProxy::tallyUnderrunFrames(). getUnderrunFrames() returns
a rolling counter for streaming tracks, or zero for static buffer tracks
which never underrun, but do a kind of 'pause' at end of buffer.
tallyUnderrunFrames() increments the counter by a specified number of frames.
Change-Id: Ib31fd73eb17cbb23888ce3af8ff29f471f5bd5a2
laybackTracks.h
hreads.cpp
racks.cpp
|
| b08ab81bab9e8cd2e3a7fff02725a387070fd078 |
31-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Add resampler comments and fix a typo"
|
| 34af02647b387a252fb02bab8e2cb9f7bd9c8abb |
30-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Add resampler comments and fix a typo
Change-Id: Ie071673875f663de4212eed4a4dff89d51a5a915
udioFlinger.cpp
udioResampler.h
hreads.cpp
hreads.h
|
| d054c32443a493513ab63529b0c8b1aca290278c |
12-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Move control block mName to createTrack() output
This is part of a series of CLs to clean up the shared memory
control block, by removing any fields that don't have to be there.
Change-Id: I6e51003a1293b6800258c31b22cff2eba42162e7
udioFlinger.cpp
udioFlinger.h
laybackTracks.h
racks.cpp
|
| 96f60d8f04432a1ed503b3e24d5736d28c63c9a2 |
12-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Rename control block flags to mFlags
Change-Id: I7b6d31e24531954ab1ecdf3ed56c19433700bd89
hreads.cpp
racks.cpp
|
| 97252bb0e40a5a9b41f020163393cf6c5da10b7e |
30-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioRecord and HAL input stream must be 16-bit PCM only"
|
| f20e1d8df84c5fbeeace0052d100982ae39bb7a4 |
12-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Rename control block server to mServer and add comments
Change-Id: Ieabd91acee92d0e84e66fbd358df5282b856306e
hreads.cpp
racks.cpp
|
| a30e75897934da2ce7b1b03bcb4b58e139d3e81e |
30-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix theoretical race using TrackBase::sampleRate()"
|
| eb144bbbd0b18427562b0d7db5b38e11f9484739 |
09-Oct-2012 |
Andrei V. FOMITCHEV <andreix.fomitchev@intel.com> |
Fix mismatched delete/new couple
In RecordThread::readInputParameters(), mRsmpInBuffer is allocated
with new[] and should be freed with delete[] instead of delete.
This error has been found by valgrind.
Change-Id: I2f947abb658a21a7d1e61eebb99270cb37d2d147
Signed-off-by: Andrei V. FOMITCHEV <andreix.fomitchev@intel.com>
Signed-off-by: David Wagner <david.wagner@intel.com>
Signed-off-by: Jack Ren <jack.ren@intel.com>
Signed-off-by: Bruce Beare <bruce.j.beare@intel.com>
hreads.cpp
|
| fa319e6d918b84f93fb5457af5d1cca6421ac517 |
30-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Remove CC_LIKELY and CC_UNLIKELY where not needed
Only keep them in performance-sensitive code
Change-Id: Ib257ddd7bc39ce9896997ffae008c524ac743d01
udioFlinger.cpp
hreads.cpp
racks.cpp
|
| 548efc94813c1dec6e8cf6c085ae41ccb04827f1 |
29-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Rename RecordThread::mInputBytes to the more generic mBufferSize
This prepares for using it in PlaybackThreads later
Change-Id: Id90a92aa6372e4b69914b0008cef07296ca5d6a3
hreads.cpp
hreads.h
|
| 9b58f63e45ef2fdfb839b9b9bb3411d81eb96128 |
16-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Move members from ThreadBase to PlaybackThread
Move mNormalFrameCount and frameCountHAL(), since they're not used by
record threads. Also comment which fields are updated by readParameters().
Change-Id: I5fc0a8a89cc637976f22d49271a5a3e136dab4e1
hreads.cpp
hreads.h
|
| 7f249fa9bcb64da324d19f551943fac7686d221c |
29-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Use standard name and type for channel mask"
|
| 175f8338e67aae6d0ca420b8f45a5f7cab4394d5 |
29-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Simplify getParameters()"
|
| f6ed423af92a56ef54bba23eba883b1f21448b54 |
16-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Treat mChannelCount as uint32_t consistently
mChannelCount was 8-, 16-, or 32-bits
Change-Id: I2cc2fedf3e33144e5c8bbd9894763282d9217f63
hreads.cpp
hreads.h
rackBase.h
|
| 2eeca1f8e18106d118f7268cb9d8597539685c5c |
29-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Move local variable declarations to point of first use"
|
| b59c1ad726b8bdf441f373d0bdc612539769334a |
29-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Declare methods in binder opcode order"
|
| 3dcd00dddec86a1c5133083ad7ba2265d49c048c |
17-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Declare methods in binder opcode order
Change-Id: I5f624b7a51ffe1a17a67c056cf984f74e4c56eac
udioFlinger.h
racks.cpp
|
| 3af48e6b28983da8c11d3d7d7fdd2088d881b462 |
29-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove obsolete mBufferEnd"
|
| b283374cbf79dc5a36f1eef0015d7baf8881792e |
29-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Move delete AudioMixer closer to point of re-allocation"
|
| 35cc4f3127322ad3e3dd1e15e8ae29ff4b4a3af6 |
25-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Remove obsolete mBufferEnd
Change-Id: I507c6109d66000bb30933ca23b912a1316f55e5e
rackBase.h
racks.cpp
|
| bd4c4fbb3b073e48963185d11a15da1fa18d2e54 |
25-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Remove obsolete TrackBase::step(), mStepCount, mStepServerFailed
Change-Id: I6347096f066b8b19451c6472db7b0671f0cf7702
rackBase.h
racks.cpp
|
| afbbc6c5f6bc470d504dcf24ceb3ac21d57fb78d |
29-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Simplify RecordTrack::stop()"
|
| 420ea49539d23ab305a7b33c0854a7a475493dd4 |
27-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "CBLK_OVERRUN, not CBLK_UNDERRUN, is for record threads"
|
| 65d30dd9ee498f7b3bfc033735458a6ff399c207 |
27-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove obsolete TrackBase::reset()"
|
| 422801c5e9b795105f28e639c4bed5ee3f4e56f3 |
26-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "HAL stream format for mixer output threads must be stereo 16-bit PCM"
|
| a8356f663014e7d4c27869629af83d8bb3441e19 |
25-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Simplify RecordTrack::stop()
Change-Id: Ib959c1e9dc9544d12277ce11bea445118b2e0521
hreads.cpp
hreads.h
racks.cpp
|
| fc38a2e0268b5e531db2975c3a81462a3593c861 |
25-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
CBLK_OVERRUN, not CBLK_UNDERRUN, is for record threads
Change-Id: I8948f76ef4717a423c37cd6ea7db4381636af612
racks.cpp
|
| 0fea74cdbc09c1259e08215e2ea90e7988d62df8 |
25-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Remove obsolete TrackBase::reset()
Change-Id: I38100b7e28a12d7af8cb40ae3f4d9cb4a0ebe701
rackBase.h
racks.cpp
|
| bfb1b832079bbb9426f72f3863199a54aefd02da |
07-Jan-2013 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: offload playback, non-blocking write
- Added specialized playback thread class for offload playback,
derived from directoutput thread.
This thread type handles specific state transitions for offloaded
tracks and offloading commands (pause/resume/drain/flush..) to audio HAL.
As opposed to other threads, does not go to standby if the track is paused.
- Added support for asynchronous write and drain operations at audio HAL.
Use a thread to handle async callback events from HAL: this avoids locking
playback thread mutex when executing the callback and cause deadlocks when
calling audio HAL functions with the playback thread mutex locked.
- Better accouting for track activity: call start/stop and release Output
methods in audio policy manager when tracks are actually added and removed
from the active tracks list.
Added a command thread in audio policy service to handle stop/release commands
asynchronously and avoid deadlocks with playback thread.
- Track terminated status is not a state anymore. This condition is othogonal
to state to permitted state transitions while terminated.
Change-Id: Id157f4b3277620568d8eace7535d9186602564de
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
udioPolicyService.h
ffects.cpp
ffects.h
laybackTracks.h
hreads.cpp
hreads.h
rackBase.h
racks.cpp
|
| 9e8fcbcd8efa51d70d1207ff57bfbfe31324287a |
25-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Move delete AudioMixer closer to point of re-allocation
No need to delete AudioMixer before readOutputParameters
Change-Id: Icafa785c9021bbe8d985e4f9527f8fc8c5e62622
hreads.cpp
|
| 7fc97ba08e2850f3f16db704b78ce78e3dbe1ff0 |
17-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
HAL stream format for mixer output threads must be stereo 16-bit PCM
Direct and tunnel output threads can support various HAL stream formats,
included encoded. But currently there are stereo 16-bit PCM assumptions
in several places for mixer and duplicating output threads:
- mMixBuffer and mixBuffer()
- AudioMixer including resampler
- FastMixer's mixBuffer
- effects
- NBAIO_Format
- anywhere FCC_2 is used
- possibly other places
Until those assumptions are removed, this CL enforces stereo 16-bit
PCM in mixer and duplicating threads at the place where the HAL format
is read. It was already being checked in checkForNewParameters_l(),
but not in readOutputParameters().
Change-Id: Ibe344cc922743da234299097aa1bb1f54795cc9b
astMixer.cpp
hreads.cpp
|
| d8ea699dc8e7dac58bb32e9cdb31b0758da25817 |
16-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Simplify getParameters()
Change-Id: Iedfeca3cd477d023c350d6d4e6eed874ee467a32
hreads.cpp
|
| 34542acfa25c6413c87a94b6f7cc315a0c496277 |
26-Jun-2013 |
Glenn Kasten <gkasten@google.com> |
Move local variable declarations to point of first use
Change-Id: Ideb83dea2c3002651c34fa646753cba598e29e93
udioFlinger.cpp
hreads.cpp
|
| fad226abd12435dbcd232f7de396f1a097b2bd5f |
17-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Use standard name and type for channel mask
Former name 'channels' was ambiguous with respect to channel count.
Change-Id: I716f792d95a7e0c787d27514ad6e93dbcef8a415
hreads.cpp
|
| 291bb6d8947c5b0c062f0895d623c529259bfa39 |
17-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
AudioRecord and HAL input stream must be 16-bit PCM only
Currently there are 16-bit PCM assumptions in several places for capture:
- resampler API
- mRsmpInBuffer and mRsmpOutBuffer
- RecordThread::threadLoop upmix, downmix, and resampling
- possibly other places
Until those assumptions are removed, this CL enforces 16-bit PCM in both
client and server at all places where a format is checked.
Change-Id: I08b0570bff626ad0d341804825a72c14e61b4233
udioFlinger.cpp
hreads.cpp
|
| 01061c1d57700b37fa879217763494cdf94d5c98 |
22-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Use AudioSystem::setLowRamDevice() to configure memory"
|
| cb155c8f4bf556b0d37c3596c4dc265ab491152b |
22-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Revert "Fix Audioflinger crash when TeeSink is enabled""
|
| 18cf609ca3f20a9da2c7e140088d7557e7479f49 |
22-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix compile warning"
|
| 1b8ae3d9f3605fab7d14c323f2118ba3c769a42a |
19-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Fix compile warning
Change-Id: I80de4a013dc65eb7c532561438fd10e005354c03
udioPolicyService.cpp
|
| 4182c4e2a07e2441fcd5c22eaff0ddfe7f826f61 |
15-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Use AudioSystem::setLowRamDevice() to configure memory
Bug: 9798886
Change-Id: I9321e3f369f1ed9429ae222e3926ebdeb012b8b0
udioFlinger.cpp
udioFlinger.h
astMixer.cpp
astMixer.h
hreads.cpp
|
| 0d61251648b5110bfc33ef5b3d19bbf65db0a7b5 |
16-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Revert "Fix Audioflinger crash when TeeSink is enabled"
This reverts commit 84e391686d7eced293913d1d7993721224ee0ba1.
Bug: 8834855
Change-Id: I8211ef5ea5d87d97ada115723df31c8057f38ca8
udioFlinger.h
|
| 153b9fe667e6e78e0218ff0159353097428c7657 |
15-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Make AudioFlinger::instantiate() more resilient when called from separate module
Bug: 8834855
Change-Id: I4cd842cdfb09d2aaaaab9df9ac3bec6179709bd3
ndroid.mk
udioFlinger.cpp
udioMixer.cpp
udioPolicyService.cpp
udioWatchdog.cpp
onfiguration.h
ffects.cpp
astMixer.cpp
astMixerState.cpp
tateQueue.cpp
tateQueueInstantiations.cpp
hreads.cpp
racks.cpp
|
| c6ae3c8a261794fd4445e4e152d1ada074a3f92f |
17-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Fix theoretical memory leak in mConfigEvents
Change-Id: I137f70676c8919661e716c33e0dd9c25c2b6285c
hreads.cpp
hreads.h
|
| ee5e8d3e8dc61f8fb1fa725278d63802074c10c5 |
17-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix indentation"
|
| 3d1982595cb9cb0856841f1f57fbdb4581189e99 |
11-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Fix indentation
Change-Id: Ia28720a7d0fad8cf110c2448b967d5648d42e017
astMixer.cpp
|
| fb1fdc9d6603aa228362e7349451f6455c9849c2 |
11-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Add comments
Change-Id: Ifbf3a46a4183c8abc0feee1c588953ab10303cc1
tateQueue.h
hreads.h
|
| 20ed5f444c149c49d0e599da09e5cec1e5cc6051 |
03-Jul-2013 |
Eric Laurent <elaurent@google.com> |
am ace2378c: am f59a4b39: fix scheduling policy service death detection
* commit 'ace2378c934b61fbd865b5ade9ff7d0df81dabe3':
fix scheduling policy service death detection
|
| ace2378c934b61fbd865b5ade9ff7d0df81dabe3 |
03-Jul-2013 |
Eric Laurent <elaurent@google.com> |
am f59a4b39: fix scheduling policy service death detection
* commit 'f59a4b393f4844c5bbc8d6212364bdddea33d232':
fix scheduling policy service death detection
|
| 1ab85ec401801ef9a9184650d0f5a1639b45eeb9 |
31-May-2013 |
Glenn Kasten <gkasten@google.com> |
Include what is needed
Remove old includes.
Header files only include other header files that they directly need themselves.
Change-Id: Ic471386808d9f42ea19ccbd59cb50a5f83a89dd0
udioFlinger.cpp
udioPolicyService.cpp
hreads.cpp
|
| f59a4b393f4844c5bbc8d6212364bdddea33d232 |
02-Jul-2013 |
Eric Laurent <elaurent@google.com> |
fix scheduling policy service death detection
Check status of transactions to scheduling policy service
and re-acquire a binder interface in case of DEAD_OBJECT.
Bug: 8875559.
Change-Id: I1e00bd44e2d4723b3ec95d5c31d9652ba08e238a
SchedulingPolicyService.cpp
chedulingPolicyService.cpp
|
| ad3af3305f024bcbbd55c894a4995e449498e1ba |
25-Mar-2013 |
Richard Fitzgerald <rf@opensource.wolfsonmicro.com> |
Public API changes for audio offload support.
NOTE: this does _not_ include all private member variables added
to classes as part of offload support. Only public/protected functions
and stubs functions/variables needed to make the changes buildable.
- isOffloadSupported() added to audio policy service
A stub implementation is required to build, this always returns false
- setParameters() added to IAudioTrack
A stub implementation is required to build, this always returns
INVALID_OPERATION
- CBlk flag for stream end
- Change AudioSystem::getRenderPosition() to take an audio_output_t
so caller can specify which output to query
- Add AudioSystem::isOffloadSupported()
This is fully implemented down to the AudioFlinger function
AudioPolicyServer::isOffloadSupported() which is just a stub
that always returns false.
- Add EVENT_STREAM_END to AudioTrack interface.
STREAM_END is used to signal when the hardware has actually finished
playing all the data it was sent.
- Add event type enumeration to media player interface AudioSink callbacks
so that the same callback can be used to handle multiple types of
event. For offloaded tracks we also have to handle STREAM_END and
TEAR_DOWN events
- Pass audio_offload_info_t to various functions used for opening outputs,
tracks and audio players. This passes additional information about the
compressed stream down to the HAL when using offload.
For publicly-available APIs this is an optional parameter (for some of
the internal and low-level APIs around the HAL interface it is mandatory)
- Add getParameters() and setParameters() API to AudioTrack
Currently dummy implementations.
- Change AudioPlayer contructor so that it takes a set of bitflags defining what
options are required. This replaces the original bool which only specified
whether to use deep buffering.
- Changes to StageFright class definition related to handling tearing-down of
an offloaded track when we need to switch back to software decode
- Define new StageFright utility functions used for offloaded tracks
Currently dummy implementations.
- AudioFlinger changes to use extended audio_config_t.
Fills in audio_offload_info_t member if this info is passed in when
opening an output.
- libvideoeditor changes required to add the new event type parameter
to AudioSink callback functions
- libmediaplayerservice changes required to add the new event type parameter
to AudioSink callback functions
Change-Id: I3ab41138aa1083d81fe83b886a9b1021ec7320f1
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Eric Laurent <elaurent@google.com>
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
udioPolicyService.h
racks.cpp
|
| 9fdcb0a9497ca290bcf364b10868587b6bde3a34 |
27-Jun-2013 |
Glenn Kasten <gkasten@google.com> |
Fix theoretical race using TrackBase::sampleRate()
In two places we assumed that TrackBase::sampleRate() would return the
same value when it is called twice in the same function. This is not
guaranteed; sampleRate() reads from the control block so the return
value could change. To fix this, only call sampleRate() once and cache
the return value to get a consistent value.
This was only a theoretical race. In MixerThread::prepareTracks_l()
it would have no bad effect. In TimedTrack::getNextBuffer() it could
cause a real problem, but we don't currently support dynamic sample rate
ratios for timed tracks.
Change-Id: I8e5c33f0121fc058d1e70c2ab5e9135397d3e0b7
hreads.cpp
racks.cpp
|
| 30755ddfd36849a0be87fee5da849f161a45a8a6 |
21-Jun-2013 |
Glenn Kasten <gkasten@android.com> |
am d3ac2fc0: am f3e21c30: am 60f3f102: Merge "Prevent AudioCommands being freed before read"
* commit 'd3ac2fc0c5429003f69d161e42bba7e94434ec09':
Prevent AudioCommands being freed before read
|
| d3ac2fc0c5429003f69d161e42bba7e94434ec09 |
21-Jun-2013 |
Glenn Kasten <gkasten@android.com> |
am f3e21c30: am 60f3f102: Merge "Prevent AudioCommands being freed before read"
* commit 'f3e21c301b8fef49eecf9c716373fcbfbaa47670':
Prevent AudioCommands being freed before read
|
| f3e21c301b8fef49eecf9c716373fcbfbaa47670 |
21-Jun-2013 |
Glenn Kasten <gkasten@android.com> |
am 60f3f102: Merge "Prevent AudioCommands being freed before read"
* commit '60f3f102ba2973cc31e5c7ea00d95bbf19968e8b':
Prevent AudioCommands being freed before read
|
| 5fa8c4bf81275d5e1f5ef71bc66fc22e3152eeb0 |
18-Jan-2013 |
Christer Fletcher <christer.fletcher@sonymobile.com> |
Prevent AudioCommands being freed before read
When AudioCommandThread::threadLoop process AudioCommands it was
possible for it to delete a command where the posting thread still
hadn't read that status from it.
If a second command signaled the thread loop to continue after it
had inserted a new command while the thread loop was actually waiting
for the first command to report that it had read the status the
thread loop would continue and delete the first command.
Changed the wait condition when waiting for the calling thread to
read status to use command->mCond instead of mWaitWorkCV. This way
it's guaranteed that the signal to continue comes from the correct
thread.
Change-Id: Ia69b48cb4fdfaf8b4c83b56a197fb9f2058a92d1
udioPolicyService.cpp
|
| 9f80dd223d83d9bb9077fb6baee056cee4eaf7e5 |
19-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
New control block for AudioTrack and AudioRecord
Main differences between old and new control block:
- removes the mutex, which was a potential source of priority inversion
- circular indices into shared buffer, which is now always a power-of-2 size
Change-Id: I4e9b7fa99858b488ac98a441fa70e31dbba1b865
udioFlinger.h
laybackTracks.h
ecordTracks.h
hreads.cpp
rackBase.h
racks.cpp
|
| fe9a901a445633e70073b1b91fca3af34d8e8132 |
12-Jun-2013 |
Eric Laurent <elaurent@google.com> |
am bb35be24: am 7be48189: Merge "audioflinger: fix effects on direct output threads" into jb-mr2-dev
* commit 'bb35be24239f05f59715960a93cd491a4cd3b4d6':
audioflinger: fix effects on direct output threads
|
| bb35be24239f05f59715960a93cd491a4cd3b4d6 |
12-Jun-2013 |
Eric Laurent <elaurent@google.com> |
am 7be48189: Merge "audioflinger: fix effects on direct output threads" into jb-mr2-dev
* commit '7be48189c0aefb82825a7689d9e1005175570779':
audioflinger: fix effects on direct output threads
|
| d0107bcd44fe608b0c00a8843d19fb6356c4cb69 |
11-Jun-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix effects on direct output threads
PlaybackThread::addTrack_l() uses the assumption that
effects are attached to a track only if the track accumulation
buffer is different from the mixer thread output buffer.
This is not true for direct output threads where only one track is active
an only one buffer is needed.
This assumption is an optimization to avoid checking for effect chains with
the same session ID each time a track is processed. The optimization
is not key if only one track is attached to the thread which is the case for
direct outputs.
Current code fails to increment the active track count in the effect chain
on direct output threads when a track is started thus making the effect
framework clear the mix buffer and produce silence each time the mixer runs.
The fix consists in removing the optimization described above.
Bug: 9324989.
Change-Id: Id7a6337450ed90d326299c2ce9fc02f4b9e2fa6f
hreads.cpp
|
| 04b035e3ccbf2919e4447c66e6483c11f2889f01 |
07-Jun-2013 |
Eric Laurent <elaurent@google.com> |
am d3a5b48a: am 18b46679: Merge "AudioFlinger: do not cache audio device when 0." into jb-mr2-dev
* commit 'd3a5b48a1bc4f5d6ff4472f54f909abaadd5fa39':
AudioFlinger: do not cache audio device when 0.
|
| d3a5b48a1bc4f5d6ff4472f54f909abaadd5fa39 |
07-Jun-2013 |
Eric Laurent <elaurent@google.com> |
am 18b46679: Merge "AudioFlinger: do not cache audio device when 0." into jb-mr2-dev
* commit '18b466790aa0a55ff4282ed8b5e550e58e0444b2':
AudioFlinger: do not cache audio device when 0.
|
| 7e1139c0377b6806942fb2a043737b3b9cf0ae91 |
07-Jun-2013 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: do not cache audio device when 0.
AudioFlinger should not cache the audio device
passed to the audio HAL when it is AUDIO_DEVICE_NONE
but keep previous valid selection instead.
Bug: 9323399.
Change-Id: I6f9480e55a21be4115453e6a5eebc5cf2536c476
ffects.cpp
hreads.cpp
|
| 7c027248e1a4ccd5b22bc4deafb03e2d87ac8f38 |
26-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Consistent whitespace
Change-Id: I118cce68d3b777f9ec9b6bfb70367496422a40f2
udioFlinger.cpp
hreads.cpp
racks.cpp
|
| 61d404efeaaf3ea1b615c160b843ac9f274d0018 |
20-May-2013 |
Dianne Hackborn <hackbod@google.com> |
Update to supply correct name for new power manager app ops.
Change-Id: Ia686319509d98a4f467479ea494e8a54f2c1e238
hreads.cpp
|
| e974c7df1c4a87ced27a773430690a20d208d40e |
15-May-2013 |
Elliott Hughes <enh@google.com> |
am effc08ed: Merge "Replace obsolete CLOCK_MONOTONIC_HR by CLOCK_MONOTONIC"
* commit 'effc08ed1eb1d55ca13997ec96f72e40a0866394':
Replace obsolete CLOCK_MONOTONIC_HR by CLOCK_MONOTONIC
|
| 8df7184c17a6f7f7fa6612b6de30af655f5eb0a7 |
10-May-2013 |
Glenn Kasten <gkasten@google.com> |
Replace obsolete CLOCK_MONOTONIC_HR by CLOCK_MONOTONIC
(cherry-pick of da1a325bc0a1421f4233e62704da4fab8b0acf7b.)
Bug: 8895727
Change-Id: I29d5dcf8519600fdd2910345449c66a5c1284646
est-resample.cpp
|
| 4a434ce11e328ab97b02ca3cc44e920bc82c198b |
10-May-2013 |
Glenn Kasten <gkasten@google.com> |
Replace obsolete CLOCK_MONOTONIC_HR by CLOCK_MONOTONIC
(cherry-pick of da1a325bc0a1421f4233e62704da4fab8b0acf7b.)
Bug: 8895727
Change-Id: I29d5dcf8519600fdd2910345449c66a5c1284646
est-resample.cpp
|
| b8ef4856ba7c0cd74ce77e0bc855f8b391f46fa7 |
10-May-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Replace obsolete CLOCK_MONOTONIC_HR by CLOCK_MONOTONIC"
|
| da1a325bc0a1421f4233e62704da4fab8b0acf7b |
10-May-2013 |
Glenn Kasten <gkasten@google.com> |
Replace obsolete CLOCK_MONOTONIC_HR by CLOCK_MONOTONIC
Bug: 8895727
Change-Id: I0abf6da941965e9e29e232943184dbc72b95d03c
est-resample.cpp
|
| e762be91c3280d837b1d48455cba90459ced7511 |
10-May-2013 |
Mathias Agopian <mathias@google.com> |
make libaudioflinger symbols visibility hidden
we export only symbols needed by clients of this library.
this saves about 130KB (1/3rd of the lib size)
Change-Id: Id81f3ecb299ee3abc0811915cf6efe87180bf15c
ndroid.mk
udioFlinger.h
udioPolicyService.h
udioResampler.h
|
| 84e391686d7eced293913d1d7993721224ee0ba1 |
07-May-2013 |
Glenn Kasten <gkasten@google.com> |
Fix Audioflinger crash when TeeSink is enabled
Bug: 8834855
Change-Id: I54665f16d79901970348a8247d9a354da2990f42
udioFlinger.h
|
| 831a0055665c3d15ff9c99ad23e5ab2b7346f2ac |
07-May-2013 |
Glenn Kasten <gkasten@google.com> |
Fix Audioflinger crash when TeeSink is enabled
Bug: 8834855
Change-Id: I54665f16d79901970348a8247d9a354da2990f42
udioFlinger.h
|
| a6560120f96685d7929d535319198485d1d5ccdf |
03-May-2013 |
Eric Laurent <elaurent@google.com> |
Merge "Remove timing jitter during startup of audio" into jb-mr2-dev
|
| 4170f515cd6148b952a07fcbe23364f5c76a2d9c |
03-May-2013 |
Eric Laurent <elaurent@google.com> |
Merge "Remove timing jitter during startup of audio" into jb-mr2-dev
|
| c114c6509b7f552e76e55a3502e926436ccfb2d8 |
25-Apr-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix duplicating track sampling rate
Add missing initialization of client proxy sampling rate
and volumes in OutputTrack constructor.
Bug: 8687522
Change-Id: I813e700315bb97083a63dd32279f1998ac775483
racks.cpp
|
| 8d2d4932b96632e9eb3af4a3d4000192ef603960 |
25-Apr-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix duplicating track sampling rate
Add missing initialization of client proxy sampling rate
and volumes in OutputTrack constructor.
Bug: 8687522
Change-Id: I813e700315bb97083a63dd32279f1998ac775483
racks.cpp
|
| f8197a6a9d9363cb52bb8a2c15c0e5a52064355e |
23-Apr-2013 |
Glenn Kasten <gkasten@google.com> |
Remove timing jitter during startup of audio
This fixes a regression introduced recently, that increased timing jitter
during the startup of the FastMixer and AudioTrack callback threads.
The regression was to make requestPriority() asynchronous as a way to
avoid an apparent priority inversion in system_server. This means that
the target thread could run briefly with the initial priority, before
the new priority takes effect.
This change removes the startup jitter for FastMixer, by making the
requestPriority() synchronous again for that case. It doesn't matter
that this restores the priority inversion involving normal mixer thread,
because it happens during startup of both threads.
The change also removes the startup jitter for the AudioTrack callback
thread, by having the target thread check whether the requestPriority()
has completed yet. If not, the target thread blocks with a timeout
until the priority boost finishes.
Finally, we now log an error message if the expected priority boost
doesn't happen.
Bug: 8698989
Change-Id: Id590e9a274b70ec1ba85b44a585ee37a22e41cbc
astMixer.cpp
SchedulingPolicyService.cpp
SchedulingPolicyService.h
chedulingPolicyService.cpp
chedulingPolicyService.h
hreads.cpp
|
| a07f17ca46db04c9d5d9e7d6b2878db59ca2b9ea |
23-Apr-2013 |
Glenn Kasten <gkasten@google.com> |
Remove timing jitter during startup of audio
This fixes a regression introduced recently, that increased timing jitter
during the startup of the FastMixer and AudioTrack callback threads.
The regression was to make requestPriority() asynchronous as a way to
avoid an apparent priority inversion in system_server. This means that
the target thread could run briefly with the initial priority, before
the new priority takes effect.
This change removes the startup jitter for FastMixer, by making the
requestPriority() synchronous again for that case. It doesn't matter
that this restores the priority inversion involving normal mixer thread,
because it happens during startup of both threads.
The change also removes the startup jitter for the AudioTrack callback
thread, by having the target thread check whether the requestPriority()
has completed yet. If not, the target thread blocks with a timeout
until the priority boost finishes.
Finally, we now log an error message if the expected priority boost
doesn't happen.
Bug: 8698989
Change-Id: Id590e9a274b70ec1ba85b44a585ee37a22e41cbc
astMixer.cpp
SchedulingPolicyService.cpp
SchedulingPolicyService.h
chedulingPolicyService.cpp
chedulingPolicyService.h
hreads.cpp
|
| 727cc2d1b378844474c775e9f42dab672d603ae9 |
16-Apr-2013 |
Glenn Kasten <gkasten@google.com> |
Use correct pid when running under test harness
This is a partial fix which addresses one of the root causes of this bug,
but only when running under test harness. It has no effect otherwise.
Bug: 8598539
Change-Id: I87444daef1d76b17544f331e651ba1c87893c381
udioFlinger.cpp
erviceUtilities.cpp
erviceUtilities.h
|
| 949a926cadbc961fbb649c91d76d7aee8ea4d7bd |
16-Apr-2013 |
Glenn Kasten <gkasten@google.com> |
Use correct pid when running under test harness
This is a partial fix which addresses one of the root causes of this bug,
but only when running under test harness. It has no effect otherwise.
Bug: 8598539
Change-Id: I87444daef1d76b17544f331e651ba1c87893c381
udioFlinger.cpp
erviceUtilities.cpp
erviceUtilities.h
|
| a75003e583b7b317951ceca13e44f3fd226012c7 |
15-Apr-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Workaround: requestPriority() is one-way" into jb-mr2-dev
|
| 5182337fcf848ca410ae61397752dfb0995067ef |
15-Apr-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Workaround: requestPriority() is one-way" into jb-mr2-dev
|
| 8d480b86ec00396b37a22803fdfe5f7bef3d8e05 |
12-Apr-2013 |
Glenn Kasten <gkasten@google.com> |
Workaround: requestPriority() is one-way
Bug: 8565696
Change-Id: If5fcdf8593e1ca15bea45217bd683b43a2a106c7
SchedulingPolicyService.cpp
|
| 9a62391fa187a345a5d1301d4bc0b165c2ea4737 |
12-Apr-2013 |
Glenn Kasten <gkasten@google.com> |
Workaround: requestPriority() is one-way
Bug: 8565696
Change-Id: If5fcdf8593e1ca15bea45217bd683b43a2a106c7
SchedulingPolicyService.cpp
|
| ecba7488c5cceaacef95836b1b476433451e7761 |
10-Apr-2013 |
Ying Wang <wangying@google.com> |
Add liblog
Bug: 8580410
Change-Id: If493d87d60d71be664ad75b140c62acadb75b0d0
ndroid.mk
|
| da0dc0af0effe9fbfb3ce3187c8472fca2baf3c6 |
10-Apr-2013 |
Ying Wang <wangying@google.com> |
Add liblog
Bug: 8580410
Change-Id: If493d87d60d71be664ad75b140c62acadb75b0d0
ndroid.mk
|
| 7fd54ff906c69400727fc8e90118a8a0429d7ab2 |
04-Apr-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: multiple tracks on direct output
Fix problems when multiple tracks exist on a direct output
thread: when multiple tracks are active, continue to update all
track states but only take into account the last track started
for audio HAL volume and mixer state control.
Bug 8388941
Change-Id: I57e6757286f41651dda99b11a5449e431812431b
hreads.cpp
|
| d595b7c858c481a07745674ce2d8a6690e980e74 |
04-Apr-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: multiple tracks on direct output
Fix problems when multiple tracks exist on a direct output
thread: when multiple tracks are active, continue to update all
track states but only take into account the last track started
for audio HAL volume and mixer state control.
Bug 8388941
Change-Id: I57e6757286f41651dda99b11a5449e431812431b
hreads.cpp
|
| d0ebb538599cc25726b856e0f5deeb8215db8c92 |
03-Apr-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: add effect config status check
Check the result of the effect engine configuration command
and do not attempt to send parameters to, enable, or process
the effect if configuration fails.
Bug 8512027
Change-Id: I8c78a05d79fba36b1a387aa5cf2700612301ac91
ffects.cpp
|
| 2a9c5cd47159c41051fe7b8366d8205a7d1d5296 |
03-Apr-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: add effect config status check
Check the result of the effect engine configuration command
and do not attempt to send parameters to, enable, or process
the effect if configuration fails.
Bug 8512027
Change-Id: I8c78a05d79fba36b1a387aa5cf2700612301ac91
ffects.cpp
|
| 75d1ef97d9799061be9631b08b56592a7cea6013 |
11-Mar-2013 |
Glenn Kasten <gkasten@google.com> |
Increase size of statistics buffer
Bug: 8354576
Change-Id: I327e9dc203e09df8abc21e589ce9056540618abd
astMixer.h
|
| 0d35f7818dedd67844cc90218d5c131a8644f802 |
11-Mar-2013 |
Glenn Kasten <gkasten@google.com> |
Increase size of statistics buffer
Bug: 8354576
Change-Id: I327e9dc203e09df8abc21e589ce9056540618abd
astMixer.h
|
| 538f498bbd5a2da51c9037a2689afc97f8bded8f |
26-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Add template class SingleStateQueue
Change-Id: If7e2bc9b2a216524ee9cbb68682e2634933b4973
ndroid.mk
|
| 5c4cc0d99d3b1cb35c5d7c237272ee53142745fb |
26-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Add template class SingleStateQueue
Change-Id: If7e2bc9b2a216524ee9cbb68682e2634933b4973
ndroid.mk
|
| a265845ecafe06c9668ce7ac67636657f4bedf86 |
26-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Miscellaneous cleanup
Abbreviation framesReady to fRdy for new systrace.
Put inline const on one line.
Use local copy of mState in state.
Improve logging.
Line length 100.
Change-Id: I8201c3ce0e53fd464fd33d02544e52c342d40b68
udioFlinger.cpp
astMixer.cpp
laybackTracks.h
hreads.cpp
rackBase.h
racks.cpp
|
| c9b2e20f7c9a71e07ef398152709c76079decbcd |
26-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Miscellaneous cleanup
Abbreviation framesReady to fRdy for new systrace.
Put inline const on one line.
Use local copy of mState in state.
Improve logging.
Line length 100.
Change-Id: I8201c3ce0e53fd464fd33d02544e52c342d40b68
udioFlinger.cpp
astMixer.cpp
laybackTracks.h
hreads.cpp
rackBase.h
racks.cpp
|
| 10774e144b3c99afad9a5070ccf50fdb12b5fb4e |
02-Mar-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "media.log cleanup" into jb-mr2-dev
|
| bf04b5860182d8f4130dcb5d6d88ee68a58c99cd |
02-Mar-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "media.log cleanup" into jb-mr2-dev
|
| d3c4b134a87b96227b90b9ec052d8a6e9880bbdf |
27-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
media.log cleanup
Remove almost all of the specific logs, but leave the media.log
logging infrastructure in place for the next time we need it.
Re-apply a few good changes that were reverted earlier:
- check logf format vs. argument list compatibility
- distinguish potentially modified and actually modified tracks in FastMixer
- fix benign bug where sq->end() was called more than once
- fix a build warning
Bug: 6490974
Change-Id: I02d3e83646c738acaebb415bd0d6b548638b4ef5
udioMixer.cpp
udioMixer.h
astMixer.cpp
hreads.cpp
hreads.h
racks.cpp
|
| ab7d72f0804fbb7e91ad9d2a16f826d97e20e5d0 |
27-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
media.log cleanup
Remove almost all of the specific logs, but leave the media.log
logging infrastructure in place for the next time we need it.
Re-apply a few good changes that were reverted earlier:
- check logf format vs. argument list compatibility
- distinguish potentially modified and actually modified tracks in FastMixer
- fix benign bug where sq->end() was called more than once
- fix a build warning
Bug: 6490974
Change-Id: I02d3e83646c738acaebb415bd0d6b548638b4ef5
udioMixer.cpp
udioMixer.h
astMixer.cpp
hreads.cpp
hreads.h
racks.cpp
|
| 5324bb2889c34cb85849ff1302215751a32c6a0d |
01-Mar-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove tee sink debugging at compile time" into jb-mr2-dev
|
| 487422017767078980bc7435f05ea2b58601b6f4 |
01-Mar-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove tee sink debugging at compile time" into jb-mr2-dev
|
| 3c3c527a792249d07938c5c6cb9b465344a22383 |
27-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Revert "Logging to investigate a crash"
This reverts commit f0ff908da019a44115109f1b4d1b6864b35a8a29.
udioMixer.cpp
|
| 409e3749a5627f1b360feb1479fcd341067a90b8 |
27-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Revert "Logging to investigate a crash"
This reverts commit f0ff908da019a44115109f1b4d1b6864b35a8a29.
udioMixer.cpp
|
| dd0bda0c47871d4964b5035a096c7c8c27fa7469 |
26-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Remove tee sink debugging at compile time
Bug: 8223560
Change-Id: Iddbfb06c45d43d9f20bb428215dd4094931e19a7
ndroid.mk
udioFlinger.cpp
udioFlinger.h
hreads.cpp
hreads.h
racks.cpp
|
| 46909e7eb074ce1b95b8a411eb71154f53f84f77 |
26-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Remove tee sink debugging at compile time
Bug: 8223560
Change-Id: Iddbfb06c45d43d9f20bb428215dd4094931e19a7
ndroid.mk
udioFlinger.cpp
udioFlinger.h
hreads.cpp
hreads.h
racks.cpp
|
| dd4abb5177b8d94cd30a787d4ce336d5110c185c |
10-Jan-2013 |
Glenn Kasten <gkasten@google.com> |
Update tee sink
Implement rotation to reduce long-term storage use.
Implement optional per-track tee.
Dynamically enable at runtime based on property, instead of at compile-time.
Dynamic frame count not yet implemented.
Bug: 8223560
Change-Id: I3706443c6ec0cb0c6656dc288715a02ad5fea63a
ndroid.mk
udioFlinger.cpp
udioFlinger.h
hreads.cpp
rackBase.h
racks.cpp
|
| da6ef1320d0161b1640dc84d7a9c5a25860c3619 |
10-Jan-2013 |
Glenn Kasten <gkasten@google.com> |
Update tee sink
Implement rotation to reduce long-term storage use.
Implement optional per-track tee.
Dynamically enable at runtime based on property, instead of at compile-time.
Dynamic frame count not yet implemented.
Bug: 8223560
Change-Id: I3706443c6ec0cb0c6656dc288715a02ad5fea63a
ndroid.mk
udioFlinger.cpp
udioFlinger.h
hreads.cpp
rackBase.h
racks.cpp
|
| 2f8025eef6c681920579f75d298a262b12a80ac9 |
18-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
A reference cannot be re-bound
Bug: 8213067
Bug: 6490974
Change-Id: If546215641fbc9a0fb1e8af8b85cc01f0a4b26ab
udioMixer.cpp
|
| fc900c922e219771d0cd3c075720d90026e1ca4a |
18-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
A reference cannot be re-bound
Bug: 8213067
Bug: 6490974
Change-Id: If546215641fbc9a0fb1e8af8b85cc01f0a4b26ab
udioMixer.cpp
|
| 5f6f3764989601d3b3ee65beaf1447d2a6b5f284 |
16-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Revert "Temporary additional logging to investigate bug"
This reverts commit 32584a7d672864b20ab8b83a3cb23c1858e908b7
Change-Id: I9dc680578b955b1af462eeb7a49d61a0d45eb81b
udioFlinger.h
udioMixer.cpp
udioMixer.h
astMixer.cpp
tateQueue.h
hreads.cpp
hreads.h
racks.cpp
|
| 7f5d335f7b4caecd0dfb8f1085f352f1d2da5d2e |
16-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Revert "Temporary additional logging to investigate bug"
This reverts commit 32584a7d672864b20ab8b83a3cb23c1858e908b7
Change-Id: I9dc680578b955b1af462eeb7a49d61a0d45eb81b
udioFlinger.h
udioMixer.cpp
udioMixer.h
astMixer.cpp
tateQueue.h
hreads.cpp
hreads.h
racks.cpp
|
| dd9764290b3c1d801fea9505189cae29db919902 |
13-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Temporary additional logging to investigate bug
The bug appears related to continuing to use an invalid buffer provider
in fast mixer after track destruction, so focus the added logs in that area.
Also includes a bug fix: was calling log in an unsafe place
near Threads.cpp AudioFlinger::PlaybackThread::createTrack_l line 1250.
Details:
- include caller pid or client pid where appropriate
- increase log buffer size
- log mFastIndex when AudioMixer sees an invalid bufferProvider.
- log both potentially modified and actually modified tracks in FastMixer.
- fix benign bug where sq->end() was called more than once.
- log StateQueue push() call and return.
- increase StateQueue size from 4 to 8 entries
- log mixer->enable(), bufferProvider, and currentTrackMask
- log buffer provider addresses
- increase fast mixer log buffer again
- check logf format vs. argument list compatibility
- add logging to AudioMixer
- add checking of magic field in AudioMixer to detect overwrites
- add bool AudioMixer::enabled()
- increase log buffer sizes yet again
- enable assertion checking without ALOGV
- improve a few log messages
- check for corruption in more places
- log in all the process hooks
- add new mixer APIs so we can check for corruption of mixer state
- fix a build warning
Bug: 6490974
Change-Id: Ib0c4a73dcf606ef9bd898313b3b40ef61ab42f51
udioFlinger.h
udioMixer.cpp
udioMixer.h
astMixer.cpp
tateQueue.h
hreads.cpp
hreads.h
racks.cpp
|
| 32584a7d672864b20ab8b83a3cb23c1858e908b7 |
13-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Temporary additional logging to investigate bug
The bug appears related to continuing to use an invalid buffer provider
in fast mixer after track destruction, so focus the added logs in that area.
Also includes a bug fix: was calling log in an unsafe place
near Threads.cpp AudioFlinger::PlaybackThread::createTrack_l line 1250.
Details:
- include caller pid or client pid where appropriate
- increase log buffer size
- log mFastIndex when AudioMixer sees an invalid bufferProvider.
- log both potentially modified and actually modified tracks in FastMixer.
- fix benign bug where sq->end() was called more than once.
- log StateQueue push() call and return.
- increase StateQueue size from 4 to 8 entries
- log mixer->enable(), bufferProvider, and currentTrackMask
- log buffer provider addresses
- increase fast mixer log buffer again
- check logf format vs. argument list compatibility
- add logging to AudioMixer
- add checking of magic field in AudioMixer to detect overwrites
- add bool AudioMixer::enabled()
- increase log buffer sizes yet again
- enable assertion checking without ALOGV
- improve a few log messages
- check for corruption in more places
- log in all the process hooks
- add new mixer APIs so we can check for corruption of mixer state
- fix a build warning
Bug: 6490974
Change-Id: Ib0c4a73dcf606ef9bd898313b3b40ef61ab42f51
udioFlinger.h
udioMixer.cpp
udioMixer.h
astMixer.cpp
tateQueue.h
hreads.cpp
hreads.h
racks.cpp
|
| ce70d400ff9fb614e334ab2b661df2933533ef57 |
15-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Revert "Temporary additional logging to investigate bug"
This reverts commit 639482c24c911b125398b31883ba6d55faebe28b
Change-Id: I11f2829072ab11e18b0663024f27bf31192f1d39
udioFlinger.h
udioMixer.cpp
udioMixer.h
astMixer.cpp
tateQueue.h
hreads.cpp
hreads.h
racks.cpp
|
| e186b51e0a9834b287d7a509e960eaf1b688db75 |
15-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Revert "Temporary additional logging to investigate bug"
This reverts commit 639482c24c911b125398b31883ba6d55faebe28b
Change-Id: I11f2829072ab11e18b0663024f27bf31192f1d39
udioFlinger.h
udioMixer.cpp
udioMixer.h
astMixer.cpp
tateQueue.h
hreads.cpp
hreads.h
racks.cpp
|
| 5881f18029deb80eb83ea88046d0593441be79c7 |
13-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Temporary additional logging to investigate bug
The bug appears related to continuing to use an invalid buffer provider
in fast mixer after track destruction, so focus the added logs in that area.
Also includes a bug fix: was calling log in an unsafe place
near Threads.cpp AudioFlinger::PlaybackThread::createTrack_l line 1250.
Details:
- include caller pid or client pid where appropriate
- increase log buffer size
- log mFastIndex when AudioMixer sees an invalid bufferProvider.
- log both potentially modified and actually modified tracks in FastMixer.
- fix benign bug where sq->end() was called more than once.
- log StateQueue push() call and return.
- increase StateQueue size from 4 to 8 entries
- log mixer->enable(), bufferProvider, and currentTrackMask
- log buffer provider addresses
- increase fast mixer log buffer again
- check logf format vs. argument list compatibility
- add logging to AudioMixer
- add checking of magic field in AudioMixer to detect overwrites
- add bool AudioMixer::enabled()
Bug: 6490974
Change-Id: I1f3f18aa62d9fbd35bc32285b669f5ba40efe28e
udioFlinger.h
udioMixer.cpp
udioMixer.h
astMixer.cpp
tateQueue.h
hreads.cpp
hreads.h
racks.cpp
|
| 639482c24c911b125398b31883ba6d55faebe28b |
13-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Temporary additional logging to investigate bug
The bug appears related to continuing to use an invalid buffer provider
in fast mixer after track destruction, so focus the added logs in that area.
Also includes a bug fix: was calling log in an unsafe place
near Threads.cpp AudioFlinger::PlaybackThread::createTrack_l line 1250.
Details:
- include caller pid or client pid where appropriate
- increase log buffer size
- log mFastIndex when AudioMixer sees an invalid bufferProvider.
- log both potentially modified and actually modified tracks in FastMixer.
- fix benign bug where sq->end() was called more than once.
- log StateQueue push() call and return.
- increase StateQueue size from 4 to 8 entries
- log mixer->enable(), bufferProvider, and currentTrackMask
- log buffer provider addresses
- increase fast mixer log buffer again
- check logf format vs. argument list compatibility
- add logging to AudioMixer
- add checking of magic field in AudioMixer to detect overwrites
- add bool AudioMixer::enabled()
Bug: 6490974
Change-Id: I1f3f18aa62d9fbd35bc32285b669f5ba40efe28e
udioFlinger.h
udioMixer.cpp
udioMixer.h
astMixer.cpp
tateQueue.h
hreads.cpp
hreads.h
racks.cpp
|
| 7f2f8042cf335ab1323dec3edbe9143a06109f4e |
14-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Revert "Temporary additional logging to investigate bug"
This reverts commit 0ddd56316262ac74a95e9edb595697c163136d6d
Change-Id: I180a928af6f5a38d15a5efe44cd1fe927b5d961c
udioFlinger.h
udioMixer.cpp
udioMixer.h
astMixer.cpp
hreads.cpp
hreads.h
racks.cpp
|
| 51eb3965caa8ba135bcdd8ffb7a2024a042ecdc0 |
14-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Revert "Temporary additional logging to investigate bug"
This reverts commit 0ddd56316262ac74a95e9edb595697c163136d6d
Change-Id: I180a928af6f5a38d15a5efe44cd1fe927b5d961c
udioFlinger.h
udioMixer.cpp
udioMixer.h
astMixer.cpp
hreads.cpp
hreads.h
racks.cpp
|
| b900234f9e772bb7591a551f06ba17ee1969368f |
13-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Temporary additional logging to investigate bug
The bug appears related to continuing to use an invalid buffer provider
in fast mixer after track destruction, so focus the added logs in that area.
Also includes a bug fix: was calling log in an unsafe place
near Threads.cpp AudioFlinger::PlaybackThread::createTrack_l line 1250.
Details:
- include caller pid or client pid where appropriate
- increase log buffer size
- log mFastIndex when AudioMixer sees an invalid bufferProvider.
- log both potentially modified and actually modified tracks in FastMixer.
- fix benign bug where sq->end() was called more than once.
- log StateQueue push() call and return.
Bug: 6490974
Change-Id: Iee7c8f40e20b6000cd8286c0ec6a14fff4a37af1
udioFlinger.h
udioMixer.cpp
udioMixer.h
astMixer.cpp
hreads.cpp
hreads.h
racks.cpp
|
| 0ddd56316262ac74a95e9edb595697c163136d6d |
13-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Temporary additional logging to investigate bug
The bug appears related to continuing to use an invalid buffer provider
in fast mixer after track destruction, so focus the added logs in that area.
Also includes a bug fix: was calling log in an unsafe place
near Threads.cpp AudioFlinger::PlaybackThread::createTrack_l line 1250.
Details:
- include caller pid or client pid where appropriate
- increase log buffer size
- log mFastIndex when AudioMixer sees an invalid bufferProvider.
- log both potentially modified and actually modified tracks in FastMixer.
- fix benign bug where sq->end() was called more than once.
- log StateQueue push() call and return.
Bug: 6490974
Change-Id: Iee7c8f40e20b6000cd8286c0ec6a14fff4a37af1
udioFlinger.h
udioMixer.cpp
udioMixer.h
astMixer.cpp
hreads.cpp
hreads.h
racks.cpp
|
| c93aff0a698265501f46393f3146b884aa357aee |
12-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Revert "Temporary additional logging to investigate bug"
This reverts commit 3051df27261e9952c0e642dec548515250e85f6a
Change-Id: I8bf5c3e91b65bd20de26f480c367c2854b62373c
udioFlinger.h
astMixer.cpp
hreads.cpp
hreads.h
racks.cpp
|
| ecd9389c8712aedeb2a79823ea0e4fb842684269 |
12-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Revert "Temporary additional logging to investigate bug"
This reverts commit 3051df27261e9952c0e642dec548515250e85f6a
Change-Id: I8bf5c3e91b65bd20de26f480c367c2854b62373c
udioFlinger.h
astMixer.cpp
hreads.cpp
hreads.h
racks.cpp
|
| a1bf28b533907b0b234e52c72bc372b3fa49c5a3 |
12-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Temporary additional logging to investigate bug
The bug appears related to continuing to use an invalid buffer provider
in fast mixer after track destruction, so focus the added logs in that area.
Also includes a bug fix: was calling log in an unsafe place
near Threads.cpp AudioFlinger::PlaybackThread::createTrack_l line 1250.
Details:
- include caller pid or client pid where appropriate
- increase log buffer size
Bug: 6490974
Change-Id: I4c030f171343fe4b483eae0ddea4427118d8d4b1
udioFlinger.h
astMixer.cpp
hreads.cpp
hreads.h
racks.cpp
|
| 3051df27261e9952c0e642dec548515250e85f6a |
12-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Temporary additional logging to investigate bug
The bug appears related to continuing to use an invalid buffer provider
in fast mixer after track destruction, so focus the added logs in that area.
Also includes a bug fix: was calling log in an unsafe place
near Threads.cpp AudioFlinger::PlaybackThread::createTrack_l line 1250.
Details:
- include caller pid or client pid where appropriate
- increase log buffer size
Bug: 6490974
Change-Id: I4c030f171343fe4b483eae0ddea4427118d8d4b1
udioFlinger.h
astMixer.cpp
hreads.cpp
hreads.h
racks.cpp
|
| e336f9113bdd35119866a3c3f8008ca01c576660 |
05-Feb-2013 |
Jean-Michel Trivi <jmtrivi@google.com> |
Add support for querying if a stream is active remotely
Bug 7485803
Change-Id: I0744374f130fd2dd0714102354cffed2fa915361
udioPolicyService.cpp
udioPolicyService.h
|
| 272ab546940054ad7991bef4b3a36f15175721cd |
05-Feb-2013 |
Jean-Michel Trivi <jmtrivi@google.com> |
Add support for querying if a stream is active remotely
Bug 7485803
Change-Id: I0744374f130fd2dd0714102354cffed2fa915361
udioPolicyService.cpp
udioPolicyService.h
|
| 83fa606bd629ac32a2dab5af255188acfa30db4a |
04-Feb-2013 |
Eric Laurent <elaurent@google.com> |
Merge "AudioFlinger: fix RecordThread initial device"
|
| 59686720182dcaa1a1a4757964df38c16f1a5246 |
04-Feb-2013 |
Eric Laurent <elaurent@google.com> |
Merge "AudioFlinger: fix RecordThread initial device"
|
| 5fa419b962f668e6fa9adf7e14fddea97cdb736e |
04-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Fix crash when destroying a DuplicatingThread
Bug: 8121343
Change-Id: I9b1fc56c753e465e8a08916022e52db6c3a36906
udioFlinger.cpp
|
| 685ef09bcaf5de6abf2064d552296f70eaec6761 |
04-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Fix crash when destroying a DuplicatingThread
Bug: 8121343
Change-Id: I9b1fc56c753e465e8a08916022e52db6c3a36906
udioFlinger.cpp
|
| 201fc9c9feb4765a12bc39e95d25f3aa1deb8ffe |
02-Feb-2013 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: fix RecordThread initial device
A regression was introduced when the audio device enums where
modified for a 32 bit representation: the device passed when
constructing a RecordThread was still the concatenation of input
device and output device bit fields on one 32 bit value which
is not possible anymore.
The fix consists in modifying the RecordThread constructor to accept
separate values for input and output devices.
Change-Id: I81fb5f4718428b54251e65d74b86e198ce15193e
udioFlinger.cpp
hreads.cpp
hreads.h
|
| d3922f72601d82c6fc067a98916fda0bd1291c5f |
02-Feb-2013 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: fix RecordThread initial device
A regression was introduced when the audio device enums where
modified for a 32 bit representation: the device passed when
constructing a RecordThread was still the concatenation of input
device and output device bit fields on one 32 bit value which
is not possible anymore.
The fix consists in modifying the RecordThread constructor to accept
separate values for input and output devices.
Change-Id: I81fb5f4718428b54251e65d74b86e198ce15193e
udioFlinger.cpp
hreads.cpp
hreads.h
|
| 011aa652a9349750dd6bca1dcb1b82f19e07504e |
19-Jan-2013 |
Glenn Kasten <gkasten@google.com> |
AudioFlinger uses media.log service for logging
Change-Id: Ia0f8204334f6b233f644d897762a18c95d936b4b
udioFlinger.cpp
udioFlinger.h
astMixer.cpp
astMixerState.cpp
astMixerState.h
hreads.cpp
hreads.h
racks.cpp
|
| 9e58b552f51b00b3b674102876bd6c77ef3da806 |
19-Jan-2013 |
Glenn Kasten <gkasten@google.com> |
AudioFlinger uses media.log service for logging
Change-Id: Ia0f8204334f6b233f644d897762a18c95d936b4b
udioFlinger.cpp
udioFlinger.h
astMixer.cpp
astMixerState.cpp
astMixerState.h
hreads.cpp
hreads.h
racks.cpp
|
| 3a948fceb6b59121cc2e5eeb458064150870ffde |
18-Jan-2013 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: fix build with debug log enabled
Change-Id: Id397155ec884c41cb84b35462ea09a97a04d3ed2
hreads.cpp
|
| 2592f6e68edbed386e004d5b045233f15e7b9ba1 |
18-Jan-2013 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: fix build with debug log enabled
Change-Id: Id397155ec884c41cb84b35462ea09a97a04d3ed2
hreads.cpp
|
| befdd0275dbb78e1a9fa2a33a3a1de731b1fe389 |
09-Jan-2013 |
Glenn Kasten <gkasten@google.com> |
Fix build warnings
Change-Id: I79f4d7052a4c0a9fde144063f44a9c8c9dd575d5
udio-resampler/AudioResamplerCoefficients.cpp
udio-resampler/filter_coefficients.h
|
| 70dc7025fac462e502803108e55911e6b3301bfe |
09-Jan-2013 |
Glenn Kasten <gkasten@google.com> |
Fix build warnings
Change-Id: I79f4d7052a4c0a9fde144063f44a9c8c9dd575d5
udio-resampler/AudioResamplerCoefficients.cpp
udio-resampler/filter_coefficients.h
|
| 856ff4e4c3c43550f013e80277358fdf514342bf |
22-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Start isolating control block accesses in a proxy"
|
| d5681bc9a38fe4cd1d591e6ae62b9c68fb851041 |
22-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Start isolating control block accesses in a proxy"
|
| 5570227ac334479982a9dd8c1647c00f58f47c18 |
21-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove unnecessary parameter"
|
| 49d24d3fcd01aef0ec1fa4abd9c1b85473f64c0f |
21-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove unnecessary parameter"
|
| 10f57df484daecc356ca1720c4ba799a8bf6761a |
20-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix build warnings"
|
| 7cd2dd3ae3278a50396886cf0b5c9a1277601816 |
20-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix build warnings"
|
| f93aa1fdae47ca2eddc7ca35fbde4266b114e8c5 |
20-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Add warning about following the design rules"
|
| cb02ba7c2666b72db030b8bf4a3f2cddc3c23ead |
20-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Add warning about following the design rules"
|
| f37971f624164c3ed185d3f976404a6f60f49b9a |
03-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Remove unnecessary parameter
Just get the parameter on server side
Change-Id: I433a63104dbb257e0d862be2ab61847cb36d1c15
udioFlinger.cpp
udioFlinger.h
|
| 8d6cc842e8d525405c68e57fdf3bc5da0b4d7e87 |
03-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Remove unnecessary parameter
Just get the parameter on server side
Change-Id: I433a63104dbb257e0d862be2ab61847cb36d1c15
udioFlinger.cpp
udioFlinger.h
|
| 92b1343f5fbc2fc43198a0a1252fe023cc9a5061 |
14-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Fix build warnings
Change-Id: I9e3cfa0c6b3467fe763ce55f759d179f02c5deea
udioResamplerSinc.cpp
udioResamplerSinc.h
hreads.cpp
|
| c4974312e5a1e2ab94eca56045f991baf1508d73 |
14-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Fix build warnings
Change-Id: I9e3cfa0c6b3467fe763ce55f759d179f02c5deea
udioResamplerSinc.cpp
udioResamplerSinc.h
hreads.cpp
|
| 552f274d82fb4988fb972df69134e031d375abfe |
04-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Start isolating control block accesses in a proxy
The proxy object will eventually be the only code that understands the
details of the control block. This should make it easier to change the
control block in the future.
Initial set of control block fields that are isolated:
- sample rate
- send level
- volume
Prepare for streaming/static separation by adding a union to the control
block for the new fields.
Fix bug in handling of max sample rate on a track. It was only checking
at re-configuration, not at each mix.
Simplify OutputTrack::obtainBuffer.
Change-Id: I2249f9d04f73a911a922ad1d7f6197292c74cd92
udioFlinger.h
laybackTracks.h
ecordTracks.h
hreads.cpp
rackBase.h
racks.cpp
|
| e3aa659e9cee7df5c12a80d285cc29ab3b2cbb39 |
04-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Start isolating control block accesses in a proxy
The proxy object will eventually be the only code that understands the
details of the control block. This should make it easier to change the
control block in the future.
Initial set of control block fields that are isolated:
- sample rate
- send level
- volume
Prepare for streaming/static separation by adding a union to the control
block for the new fields.
Fix bug in handling of max sample rate on a track. It was only checking
at re-configuration, not at each mix.
Simplify OutputTrack::obtainBuffer.
Change-Id: I2249f9d04f73a911a922ad1d7f6197292c74cd92
udioFlinger.h
laybackTracks.h
ecordTracks.h
hreads.cpp
rackBase.h
racks.cpp
|
| 30c01816ddee2268fcd3d965240cba9c00301e7d |
04-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Prepare for track invalidation to be done by proxy
Don't rely on control block to determine whether track has been marked
invalid. Instead, use a local flag that can't be corrupted by client.
Change-Id: I783dafe828f93c1c3d2d0e5a08105ea536436efb
laybackTracks.h
hreads.cpp
racks.cpp
|
| 5736c35b841de56ce394b4879389f669b61425e6 |
04-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Prepare for track invalidation to be done by proxy
Don't rely on control block to determine whether track has been marked
invalid. Instead, use a local flag that can't be corrupted by client.
Change-Id: I783dafe828f93c1c3d2d0e5a08105ea536436efb
laybackTracks.h
hreads.cpp
racks.cpp
|
| e1cf3c41bd27b5ab7d8e70ef8dd6fcd0f5453568 |
11-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
am 76143093: am f0ff908d: Logging to investigate a crash
* commit '761430930a7335d714af0f7c1d497feb8f9f0a5a':
Logging to investigate a crash
|
| ae162976dda428671af09a8fbc3f03173a7e6f3e |
11-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
am 76143093: am f0ff908d: Logging to investigate a crash
* commit '761430930a7335d714af0f7c1d497feb8f9f0a5a':
Logging to investigate a crash
|
| f0ff908da019a44115109f1b4d1b6864b35a8a29 |
07-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Logging to investigate a crash
Bug: 6490974
Change-Id: Ib926a9258bde4ee05ed42eea662dff68e426a997
udioMixer.cpp
|
| ef5abc369823f634c460c8f24d3cbf310542fe3b |
07-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Logging to investigate a crash
Bug: 6490974
Change-Id: Ib926a9258bde4ee05ed42eea662dff68e426a997
udioMixer.cpp
|
| e0da485dd76267e86cff7cbd334c74435d745225 |
04-Dec-2012 |
Alex Ray <aray@google.com> |
Merge "audioflinger: define ANDROID_SMP, remove conditional tracing"
|
| 811cddad95a75c71dfeca0965c92f710c81e192f |
04-Dec-2012 |
Alex Ray <aray@google.com> |
Merge "audioflinger: define ANDROID_SMP, remove conditional tracing"
|
| 4b3a49e3a9ced6e63adab08acc5b720f8feddefa |
29-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
AudioTrack::mute() is unused so remove it
If ever needed again, it could be implemented on client side by forcing
a track volume of 0.
Change-Id: I88a9b4f675b6dca2948549414f9ec2c192d29269
udioFlinger.h
laybackTracks.h
hreads.cpp
racks.cpp
|
| e4756fe3a387615acb63c6a05788c8db9b5786cb |
29-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
AudioTrack::mute() is unused so remove it
If ever needed again, it could be implemented on client side by forcing
a track volume of 0.
Change-Id: I88a9b4f675b6dca2948549414f9ec2c192d29269
udioFlinger.h
laybackTracks.h
hreads.cpp
racks.cpp
|
| 71b6e60c52ce367bc0d64ffca60848d600511b65 |
01-Dec-2012 |
Alex Ray <aray@google.com> |
audioflinger: define ANDROID_SMP, remove conditional tracing
With ANDROID_SMP set, tracing functionality is completely inline, and
without the performance hits of external library calls, tracing does
not need to be conditionally compiled.
Change-Id: I4b29a9a52c403f0d2ea137c5b7bc05a518a7ca4b
ndroid.mk
astMixer.cpp
|
| b3a8364eeea621ef63b983e4c1b0771f62069fe0 |
01-Dec-2012 |
Alex Ray <aray@google.com> |
audioflinger: define ANDROID_SMP, remove conditional tracing
With ANDROID_SMP set, tracing functionality is completely inline, and
without the performance hits of external library calls, tracing does
not need to be conditionally compiled.
Change-Id: I4b29a9a52c403f0d2ea137c5b7bc05a518a7ca4b
ndroid.mk
astMixer.cpp
|
| af348741539ae4a84aa929a54ecf359dae96f3ee |
30-Nov-2012 |
Alex Ray <aray@google.com> |
Remove conditional compilation of ATRACE functions
Tracing functions are meant to be dynamically controlled via sysprops.
Conditional compilation removes this functionality.
Change-Id: I26bc473d104d0b3c50a228dddfda3fa2428d157a
ndroid.mk
astMixer.cpp
hreads.cpp
|
| 371eb9756c32109ea572b91216b19bb623f6d3fd |
30-Nov-2012 |
Alex Ray <aray@google.com> |
Remove conditional compilation of ATRACE functions
Tracing functions are meant to be dynamically controlled via sysprops.
Conditional compilation removes this functionality.
Change-Id: I26bc473d104d0b3c50a228dddfda3fa2428d157a
ndroid.mk
astMixer.cpp
hreads.cpp
|
| 05f9e4fe9a97e077b5ed37214464c0a67af80871 |
30-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Add warning about following the design rules
Change-Id: Ic4895ed5682bad10b03e97d8015e642ee1696533
astMixer.cpp
|
| a3d2628a22f2b3d682495044897a40ea1399a662 |
30-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Add warning about following the design rules
Change-Id: Ic4895ed5682bad10b03e97d8015e642ee1696533
astMixer.cpp
|
| e8cb004257565111cb261f3e9c786adba881859a |
30-Nov-2012 |
Simon Wilson <simonwilson@google.com> |
Merge "Use ATRACE macros instead of Tracer statics"
|
| 45f7c446777171a728f9eb257f48145dde5d19e2 |
30-Nov-2012 |
Simon Wilson <simonwilson@google.com> |
Merge "Use ATRACE macros instead of Tracer statics"
|
| 7a90bc9265782788675af577c7b1c56e5d5be709 |
30-Nov-2012 |
Simon Wilson <simonwilson@google.com> |
Use ATRACE macros instead of Tracer statics
ATRACE_BEGIN and ATRACE_END have replaced the static
Tracer::traceBegin and Tracer::traceEnd functions, so
use them instead.
Fixes compilation errors when tracing is enabled.
Change-Id: I4d1147d2f76afcdf113e9986f0544cb848802b15
astMixer.cpp
hreads.cpp
|
| 2d590964aa58e137d17a43e095e6443dd0fe2e98 |
30-Nov-2012 |
Simon Wilson <simonwilson@google.com> |
Use ATRACE macros instead of Tracer statics
ATRACE_BEGIN and ATRACE_END have replaced the static
Tracer::traceBegin and Tracer::traceEnd functions, so
use them instead.
Fixes compilation errors when tracing is enabled.
Change-Id: I4d1147d2f76afcdf113e9986f0544cb848802b15
astMixer.cpp
hreads.cpp
|
| f15a6227840b8d04a70fbd45fbd4de06b21fdc1f |
29-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
am 9428cdd7: am 5dea84c3: Fix log spam
* commit '9428cdd7b9d32647d35207b3579133ee6a1f7355':
Fix log spam
|
| 8fac1ce6bd56e3a4b0c26c1c177bf76d9ee19da6 |
29-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
am 9428cdd7: am 5dea84c3: Fix log spam
* commit '9428cdd7b9d32647d35207b3579133ee6a1f7355':
Fix log spam
|
| 5dea84c39b5e790315b525009c635806e33c97e0 |
29-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Fix log spam
Change-Id: Ie6c982af906dcfd3cdea4b771dfab1f7e47745ca
udio-resampler/AudioResamplerCoefficients.cpp
|
| 65ac1b1761c57124382355c21edbbdaaab2ee025 |
29-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Fix log spam
Change-Id: Ie6c982af906dcfd3cdea4b771dfab1f7e47745ca
udio-resampler/AudioResamplerCoefficients.cpp
|
| 9bcb476a95a26e62f5706d1f00f4873cf44f9e04 |
19-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
New VHQ resampler
Squashed commit of the following:
commit 12b6952da9f25e94d06dd7185bce255924e7e791
Author: Mathias Agopian <mathias@google.com>
Date: Mon Nov 19 15:27:26 2012 -0800
fix a typo in SINC resampler that prevented tracks to be mixed
we were always erasing the current mix instead of mixing into it.
Change-Id: Ib229245f9e5a0d384f1727640a59e9f0469211a2
commit 0019ce082df430278f14ab922e900ce33b64897d
Author: Dave Bort <dbort@google.com>
Date: Tue Nov 13 01:30:32 2007 -0800
Rename "TARGET" to "MODULE" in the build system.
Part one of the grand renaming.
API_CHANGE: Third parties may need to update their makefiles.
Any variables with "LOCAL" and "TARGET" in their names
should now use "MODULE" instead of "TARGET"; e.g., LOCAL_MODULE,
LOCAL_MODULE_TAGS.
PRESUBMIT=passed
OCL=39840
Change-Id: Ica9a7937d3d9552ab84db46ac6eea8a290e404fe
Signed-off-by: Glenn Kasten <gkasten@google.com>
commit f01adc0cef0e39e75c76d9195ac26a94cac0a100
Author: Glenn Kasten <gkasten@google.com>
Date: Wed Nov 14 08:32:08 2012 -0800
Fix build warnings
Change-Id: Ic43bcca166a529a6431711b05a7fa21849b6a38b
commit 9bb031a565c753a03d9c9397edea318947d80528
Author: Mathias Agopian <mathias@google.com>
Date: Sat Nov 10 04:44:30 2012 -0800
more optimizations...
calculate the offsets from the phase differently, this happens
to reduce the register pressure in the main loop, which in turns
allows the compiler to generate much better code (doesn't need
to spill a lot of stuff on the stack).
this gives another 15% performance increase
Change-Id: I2ce3479dd48b9e6941adb80e6d443d6e14d64d96
commit 5a951598f31217b8cd2babd0720c9608ee17291a
Author: Mathias Agopian <mathias@google.com>
Date: Sat Nov 10 03:26:39 2012 -0800
refactor code to improve neon code
we want to make sure we don't transfer data from the
neon unit to the arm register file, as this can be quite
slow. instead we do all the calculation on the neon side
and write the result directly to main memory.
Change-Id: Ibb56664d3ab03098ae2798b75e2b6927ac900187
commit b381ee9e83bc9fd18986e79c7809841514ed590e
Author: Mathias Agopian <mathias@google.com>
Date: Sun Nov 4 15:16:13 2012 -0800
NEON optimized SINC resampler
this currently gives us a 60% to 80% boost depending
on the quality level selected.
Change-Id: I7db385007e811ed7bffe5fd3403b44e300894f5b
commit bea077354210242ea193a50b0dbab0fedab25df3
Author: Mathias Agopian <mathias@google.com>
Date: Mon Nov 5 01:51:37 2012 -0800
minor cleanups
Change-Id: Ia12ee4fb59e90221761bec85e6450db29197591f
commit 8f4ed7decbe161a5ff38200b218f5216d80aba46
Author: Mathias Agopian <mathias@google.com>
Date: Sun Nov 4 18:49:14 2012 -0800
improve resample test
- handle stereo input
- input file can now be ommited, in this case
a linear chirp will be used automatically
- better usage information
Change-Id: I5d62a6c26a9054a1c1a517a065b4df5a2cdcda22
commit 5fcd634ea6cb4df27c495abe20f5f9b8ff55d128
Author: Mathias Agopian <mathias@google.com>
Date: Sun Nov 4 02:03:49 2012 -0800
change how we store the FIR coefficients
The coefficient table is now transposed and shows
much better its polyphase nature: we now have a FIR
per line, each line corresponding to a phase.
This doesn't change at all the results produced by
the filter, but allows us to make slightly better
use of the data cache and improves performance a bit
(although not as much as I thought it would).
The main benefit is that it is the first step
before we can make much larger optimizations
(like using NEON).
Change-Id: Iebf7695825dcbd41f25861efcaefbaa3365ecb43
commit d652231abf4c7e2ea1fc89caae730cec1f7259a1
Author: Mathias Agopian <mathias@google.com>
Date: Sat Nov 3 23:37:53 2012 -0700
improve SINC resampler performance
The improvement is about 60% by just tweaking a few
things to help the compiler generate better code.
It turns out that inlining too much stuff manually was hurting us.
Change-Id: I8068f0f75051f95ac600e50ce552572dd1e8c304
commit 9dc68ef5b94c700c4ee68790e8cbb334c90a538d
Author: Mathias Agopian <mathias@google.com>
Date: Thu Nov 1 21:03:46 2012 -0700
new coefficients for the vhq resampler
previous coefficients were provided by a 3rd party and didn't have a
way to re-generate them. we're now using the 'fir' utility.
the performance of the filter is virtually identical, except for
the down-sampling case which seems slightly better now:
It looks like both the previous and new coefficients are generating
some sort of clipping for full-scale signals in the down-sampling case
(although the new ones seem better), the reason for that is
unknown (see bug: 7453062)
Also updated the HQ coefficients for the down-samplers, previous ones
were a little bit too conservative -- the new ones push the cut-off
frequency up by about 1 KHz.
Change-Id: I54a827b5c707c7cc41268ed01283758dce1d7647
commit 38e0b8560a6fc1b7124e22e0e09a84a285182f8e
Author: Mathias Agopian <mathias@google.com>
Date: Tue Oct 30 13:51:44 2012 -0700
fix SINC resampler on non ARM architectures
make sure the C version of the code generates the same
output than the ARM assemply version.
Change-Id: Ide218785c35d02598b2d7278e646b1b178148698
commit a1878128b182696ba508569b4d211d0dfae92463
Author: Mathias Agopian <mathias@google.com>
Date: Tue Oct 30 12:49:07 2012 -0700
fix another issue with generating FIR coefficients
the impulse response of a low-pass is 2*f*sinc(2*pi*f*k), we were
missing the 2*f scale factor. This explains why we were seeing
clipping and had to manually scale the filter down.
Change-Id: I86d0bb82ecdd99681c8ba5a8112a8257bf6f0186
commit 1a0fb993430acc9f601e6c538305bc407c20ac5d
Author: Mathias Agopian <mathias@google.com>
Date: Mon Oct 29 17:13:20 2012 -0700
fir a typo that caused up-sampling coefficiens to be wrong
up-sample coefficient were generated with a cut-off frequency of 24KHz
intead of ~20KHz, which caused more aliasing in the audible band.
also increased the attenuation to 1.3 dB on both up and down
sampling coefficient to avoid clipping.
Change-Id: Ie8aeecf1429190541b656810c6716b6aae5ece2e
commit 9520ad6862bd682ad075a9d9e3e94ada9f6e58b6
Author: Mathias Agopian <mathias@google.com>
Date: Mon Oct 29 17:13:16 2012 -0700
test-resample: clip instead of overflowing
Change-Id: I550e5a59e51c11e1095ca338222b094f92b96878
commit ba36656300f250f7f1fdeb75149749344260e6cb
Author: Mathias Agopian <mathias@google.com>
Date: Sun Oct 21 01:01:38 2012 -0700
a test app for the resamplers
Change-Id: I66852d90d384f1d9e77b51ad1a1ebdbaf61d0607
commit 056a08b9bfd33cf27228c992adc8293a56b01be8
Author: Mathias Agopian <mathias@google.com>
Date: Fri Oct 26 14:11:01 2012 -0700
reenable the cubic resampler
cubic resampler was disabled because it hadn't been qualified,
however after I did some tests, it does improve significantly
the sound quality over the order-1 resampler, even if it is
still quite bad.
also HIGH_QUALITY resampler was partially disabled, it's now
fully enabled. It's a big improvement over the cubic resampler
in terms of aliasing noise (it's not as good in the pass-band).
Change-Id: I70e3658c255896588642697be9eb594ff4ec0f8b
commit 8c0241d3ff50ae85167f69b3bd369244894cfa44
Author: Mathias Agopian <mathias@google.com>
Date: Fri Oct 26 13:48:42 2012 -0700
improve SINC resampler coefficients
- we increase the interpolation precision from 4 to 7 bits
this doesn't increase CPU power required, it only increases the
size of the filter table but significantly reduces the noise
introduced by the quantization of the impulse response.
- the parameters of the filter are set such that aliasing is
rejected at 80 dB below 20 KHz. Because we don't use a lot of
coefficient (to save compute power), there are quite a bit of
attenuation in the pass-band: starting at 9KHz for the
down-sampler (48 to 44.1), and starting at 13 KHz for the
up-sampler (44.1 to 48) -- the transition band is about 15 KHz.
Change-Id: I855548d2aab8a0fb0d2a2da3a364b6842d7d3838
commit 69e7dab2192adc1f780464146810629ebd01b145
Author: Pixelflinger <mathias.agopian@gmail.com>
Date: Thu Oct 25 19:43:49 2012 -0700
improve fir tool: cleanup, better default, bug fixes
- all parameters can be changed on the command-line
- added float output
- added debug option
- added an option to generate a polyphase filter coefficients
- added an attenuation option in dBFS
- added a lot of comments and references
- fixed kaiser window parameter
also the default should generate a filter with 80 dB rejection
(of the 24 KHz aliasing) above 20 KHz and a 15 KHz transition
band around ~20 KHz (for 48 KHz sampling rate).
It's not very good but corresponds to the current code.
commit 8347499d105a50257c18e9dac652e750b06428b1
Author: Glenn Kasten <gkasten@google.com>
Date: Mon Oct 22 17:09:27 2012 -0700
Increase allowed number of VHQ resamplers to 3
Bug: 7378660
Change-Id: I69e33ca2eb4bb9bd38e2c63df62cd1130d68baf6
commit f91cf3cad7f5c4d52614271c89ab468741c5d24c
Author: Mathias Agopian <mathias@google.com>
Date: Sun Oct 21 03:04:05 2012 -0700
Fix a typo that caused the high quality resampler to produce garbage
the problem is that if libaudio_resampler is present, it is those
coefficients that will always be selected, but the correct
meta-data.
Bug: 7385994
Change-Id: Ieebeb37b4dfb62a1a051bc29fae2ce056dbc6621
commit e158a9e4262a174c59469a205658bc3ca4078234
Author: Dan Bornstein <danfuzz@google.com>
Date: Fri Oct 3 10:34:57 2008 -0700
Manually merge change #111620 from tc3 to mainline, to keep the
automerger from choking on it.
p4 sync
p4 integrate -r -b android_to_tc3 //...@111620,111620
p4 resolve -a
p4 resolve # resolve a couple merge travesties
PRESUBMIT=passed
BUG=1399648
TBR=edheyl
OCL=111902
Change-Id: I854b01553dd92bbf9c864f5a9bd51a3d665f0ac2
Signed-off-by: Glenn Kasten <gkasten@google.com>
commit b9f3c26032be7a6ea01a10d93d94826f449e68ab
Author: Dave Bort <dbort@google.com>
Date: Fri Jan 18 14:51:05 2008 -0800
Rename "Makefile" to "Android.mk" throughout the tree.
For <http://b/issue?id=960416>.
I've tested this as much as I can, but 1500 open files =
easy to mess things up. Please let me know if there's
a problem rather than rolling back this change.
PRESUBMIT=passed
BUG=960416
TBR=joeo
OCL=46537
Change-Id: I5a404caf0f398a7afa7ae7abaf2f2a1c6ab490eb
Signed-off-by: Glenn Kasten <gkasten@google.com>
commit 0c22a9a44c4103483fba1d944acf1354c5eb1617
Author: Mathias Agopian <mathias@google.com>
Date: Mon Oct 29 23:44:25 2007 -0700
Tweak the SINC resampler parameters and double the performance. It's using about 10% CPU in the worse case now.
Change-Id: I50ac7e6c6702a427fa36ab6d976c507155057507
Signed-off-by: Glenn Kasten <gkasten@google.com>
commit b85e41487983ad085b859acf8251e7e54480308a
Author: Mathias Agopian <mathias@google.com>
Date: Mon Oct 29 04:34:36 2007 -0700
A sinc resampler for Audioflinger. It's not enabled yet, but fully functional and apparently working. It need more "quality" tests. In the 48->44 KHz, it takes about 25% of the CPU time.
Change-Id: I80eb5185e13ebdb907e0b85c49ba1272c23d60ec
Signed-off-by: Glenn Kasten <gkasten@google.com>
commit ba3949ef17cac2ba71cc3096c413782a49c922e5
Author: Mathias Agopian <mathias@google.com>
Date: Thu Aug 23 21:01:28 2007 -0700
fix a few small typos in the FIR computation
Change-Id: I6e56b514fe520f30f7487f85c64ea5d2a7c19b40
Signed-off-by: Glenn Kasten <gkasten@google.com>
commit 7474bfa7de2604021963794dddfe44985648db6a
Author: Mathias Agopian <mathias@google.com>
Date: Thu Aug 23 03:16:02 2007 -0700
This is a tool to compute the the reconstruction filter coefficients for a sinc audio resampler.
Change-Id: I99be2505139b8e0e7647200e1647509d4f7e6067
Signed-off-by: Glenn Kasten <gkasten@google.com>
Bug: 7577965
Change-Id: I2c84a9283a1668723bad83e1a119c849c88c3e6b
ndroid.mk
udioResampler.cpp
udioResamplerSinc.cpp
udioResamplerSinc.h
udio-resampler/AudioResamplerCoefficients.cpp
udio-resampler/dnsampler_filter_coefficients_x128_10112011.h
udio-resampler/filter_coefficients.h
udio-resampler/resampler_filter_coefficients_10042011.h
est-resample.cpp
|
| a614426bc7e195af227a585bb0a07f43b4022d78 |
19-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
New VHQ resampler
Squashed commit of the following:
commit 12b6952da9f25e94d06dd7185bce255924e7e791
Author: Mathias Agopian <mathias@google.com>
Date: Mon Nov 19 15:27:26 2012 -0800
fix a typo in SINC resampler that prevented tracks to be mixed
we were always erasing the current mix instead of mixing into it.
Change-Id: Ib229245f9e5a0d384f1727640a59e9f0469211a2
commit 0019ce082df430278f14ab922e900ce33b64897d
Author: Dave Bort <dbort@google.com>
Date: Tue Nov 13 01:30:32 2007 -0800
Rename "TARGET" to "MODULE" in the build system.
Part one of the grand renaming.
API_CHANGE: Third parties may need to update their makefiles.
Any variables with "LOCAL" and "TARGET" in their names
should now use "MODULE" instead of "TARGET"; e.g., LOCAL_MODULE,
LOCAL_MODULE_TAGS.
PRESUBMIT=passed
OCL=39840
Change-Id: Ica9a7937d3d9552ab84db46ac6eea8a290e404fe
Signed-off-by: Glenn Kasten <gkasten@google.com>
commit f01adc0cef0e39e75c76d9195ac26a94cac0a100
Author: Glenn Kasten <gkasten@google.com>
Date: Wed Nov 14 08:32:08 2012 -0800
Fix build warnings
Change-Id: Ic43bcca166a529a6431711b05a7fa21849b6a38b
commit 9bb031a565c753a03d9c9397edea318947d80528
Author: Mathias Agopian <mathias@google.com>
Date: Sat Nov 10 04:44:30 2012 -0800
more optimizations...
calculate the offsets from the phase differently, this happens
to reduce the register pressure in the main loop, which in turns
allows the compiler to generate much better code (doesn't need
to spill a lot of stuff on the stack).
this gives another 15% performance increase
Change-Id: I2ce3479dd48b9e6941adb80e6d443d6e14d64d96
commit 5a951598f31217b8cd2babd0720c9608ee17291a
Author: Mathias Agopian <mathias@google.com>
Date: Sat Nov 10 03:26:39 2012 -0800
refactor code to improve neon code
we want to make sure we don't transfer data from the
neon unit to the arm register file, as this can be quite
slow. instead we do all the calculation on the neon side
and write the result directly to main memory.
Change-Id: Ibb56664d3ab03098ae2798b75e2b6927ac900187
commit b381ee9e83bc9fd18986e79c7809841514ed590e
Author: Mathias Agopian <mathias@google.com>
Date: Sun Nov 4 15:16:13 2012 -0800
NEON optimized SINC resampler
this currently gives us a 60% to 80% boost depending
on the quality level selected.
Change-Id: I7db385007e811ed7bffe5fd3403b44e300894f5b
commit bea077354210242ea193a50b0dbab0fedab25df3
Author: Mathias Agopian <mathias@google.com>
Date: Mon Nov 5 01:51:37 2012 -0800
minor cleanups
Change-Id: Ia12ee4fb59e90221761bec85e6450db29197591f
commit 8f4ed7decbe161a5ff38200b218f5216d80aba46
Author: Mathias Agopian <mathias@google.com>
Date: Sun Nov 4 18:49:14 2012 -0800
improve resample test
- handle stereo input
- input file can now be ommited, in this case
a linear chirp will be used automatically
- better usage information
Change-Id: I5d62a6c26a9054a1c1a517a065b4df5a2cdcda22
commit 5fcd634ea6cb4df27c495abe20f5f9b8ff55d128
Author: Mathias Agopian <mathias@google.com>
Date: Sun Nov 4 02:03:49 2012 -0800
change how we store the FIR coefficients
The coefficient table is now transposed and shows
much better its polyphase nature: we now have a FIR
per line, each line corresponding to a phase.
This doesn't change at all the results produced by
the filter, but allows us to make slightly better
use of the data cache and improves performance a bit
(although not as much as I thought it would).
The main benefit is that it is the first step
before we can make much larger optimizations
(like using NEON).
Change-Id: Iebf7695825dcbd41f25861efcaefbaa3365ecb43
commit d652231abf4c7e2ea1fc89caae730cec1f7259a1
Author: Mathias Agopian <mathias@google.com>
Date: Sat Nov 3 23:37:53 2012 -0700
improve SINC resampler performance
The improvement is about 60% by just tweaking a few
things to help the compiler generate better code.
It turns out that inlining too much stuff manually was hurting us.
Change-Id: I8068f0f75051f95ac600e50ce552572dd1e8c304
commit 9dc68ef5b94c700c4ee68790e8cbb334c90a538d
Author: Mathias Agopian <mathias@google.com>
Date: Thu Nov 1 21:03:46 2012 -0700
new coefficients for the vhq resampler
previous coefficients were provided by a 3rd party and didn't have a
way to re-generate them. we're now using the 'fir' utility.
the performance of the filter is virtually identical, except for
the down-sampling case which seems slightly better now:
It looks like both the previous and new coefficients are generating
some sort of clipping for full-scale signals in the down-sampling case
(although the new ones seem better), the reason for that is
unknown (see bug: 7453062)
Also updated the HQ coefficients for the down-samplers, previous ones
were a little bit too conservative -- the new ones push the cut-off
frequency up by about 1 KHz.
Change-Id: I54a827b5c707c7cc41268ed01283758dce1d7647
commit 38e0b8560a6fc1b7124e22e0e09a84a285182f8e
Author: Mathias Agopian <mathias@google.com>
Date: Tue Oct 30 13:51:44 2012 -0700
fix SINC resampler on non ARM architectures
make sure the C version of the code generates the same
output than the ARM assemply version.
Change-Id: Ide218785c35d02598b2d7278e646b1b178148698
commit a1878128b182696ba508569b4d211d0dfae92463
Author: Mathias Agopian <mathias@google.com>
Date: Tue Oct 30 12:49:07 2012 -0700
fix another issue with generating FIR coefficients
the impulse response of a low-pass is 2*f*sinc(2*pi*f*k), we were
missing the 2*f scale factor. This explains why we were seeing
clipping and had to manually scale the filter down.
Change-Id: I86d0bb82ecdd99681c8ba5a8112a8257bf6f0186
commit 1a0fb993430acc9f601e6c538305bc407c20ac5d
Author: Mathias Agopian <mathias@google.com>
Date: Mon Oct 29 17:13:20 2012 -0700
fir a typo that caused up-sampling coefficiens to be wrong
up-sample coefficient were generated with a cut-off frequency of 24KHz
intead of ~20KHz, which caused more aliasing in the audible band.
also increased the attenuation to 1.3 dB on both up and down
sampling coefficient to avoid clipping.
Change-Id: Ie8aeecf1429190541b656810c6716b6aae5ece2e
commit 9520ad6862bd682ad075a9d9e3e94ada9f6e58b6
Author: Mathias Agopian <mathias@google.com>
Date: Mon Oct 29 17:13:16 2012 -0700
test-resample: clip instead of overflowing
Change-Id: I550e5a59e51c11e1095ca338222b094f92b96878
commit ba36656300f250f7f1fdeb75149749344260e6cb
Author: Mathias Agopian <mathias@google.com>
Date: Sun Oct 21 01:01:38 2012 -0700
a test app for the resamplers
Change-Id: I66852d90d384f1d9e77b51ad1a1ebdbaf61d0607
commit 056a08b9bfd33cf27228c992adc8293a56b01be8
Author: Mathias Agopian <mathias@google.com>
Date: Fri Oct 26 14:11:01 2012 -0700
reenable the cubic resampler
cubic resampler was disabled because it hadn't been qualified,
however after I did some tests, it does improve significantly
the sound quality over the order-1 resampler, even if it is
still quite bad.
also HIGH_QUALITY resampler was partially disabled, it's now
fully enabled. It's a big improvement over the cubic resampler
in terms of aliasing noise (it's not as good in the pass-band).
Change-Id: I70e3658c255896588642697be9eb594ff4ec0f8b
commit 8c0241d3ff50ae85167f69b3bd369244894cfa44
Author: Mathias Agopian <mathias@google.com>
Date: Fri Oct 26 13:48:42 2012 -0700
improve SINC resampler coefficients
- we increase the interpolation precision from 4 to 7 bits
this doesn't increase CPU power required, it only increases the
size of the filter table but significantly reduces the noise
introduced by the quantization of the impulse response.
- the parameters of the filter are set such that aliasing is
rejected at 80 dB below 20 KHz. Because we don't use a lot of
coefficient (to save compute power), there are quite a bit of
attenuation in the pass-band: starting at 9KHz for the
down-sampler (48 to 44.1), and starting at 13 KHz for the
up-sampler (44.1 to 48) -- the transition band is about 15 KHz.
Change-Id: I855548d2aab8a0fb0d2a2da3a364b6842d7d3838
commit 69e7dab2192adc1f780464146810629ebd01b145
Author: Pixelflinger <mathias.agopian@gmail.com>
Date: Thu Oct 25 19:43:49 2012 -0700
improve fir tool: cleanup, better default, bug fixes
- all parameters can be changed on the command-line
- added float output
- added debug option
- added an option to generate a polyphase filter coefficients
- added an attenuation option in dBFS
- added a lot of comments and references
- fixed kaiser window parameter
also the default should generate a filter with 80 dB rejection
(of the 24 KHz aliasing) above 20 KHz and a 15 KHz transition
band around ~20 KHz (for 48 KHz sampling rate).
It's not very good but corresponds to the current code.
commit 8347499d105a50257c18e9dac652e750b06428b1
Author: Glenn Kasten <gkasten@google.com>
Date: Mon Oct 22 17:09:27 2012 -0700
Increase allowed number of VHQ resamplers to 3
Bug: 7378660
Change-Id: I69e33ca2eb4bb9bd38e2c63df62cd1130d68baf6
commit f91cf3cad7f5c4d52614271c89ab468741c5d24c
Author: Mathias Agopian <mathias@google.com>
Date: Sun Oct 21 03:04:05 2012 -0700
Fix a typo that caused the high quality resampler to produce garbage
the problem is that if libaudio_resampler is present, it is those
coefficients that will always be selected, but the correct
meta-data.
Bug: 7385994
Change-Id: Ieebeb37b4dfb62a1a051bc29fae2ce056dbc6621
commit e158a9e4262a174c59469a205658bc3ca4078234
Author: Dan Bornstein <danfuzz@google.com>
Date: Fri Oct 3 10:34:57 2008 -0700
Manually merge change #111620 from tc3 to mainline, to keep the
automerger from choking on it.
p4 sync
p4 integrate -r -b android_to_tc3 //...@111620,111620
p4 resolve -a
p4 resolve # resolve a couple merge travesties
PRESUBMIT=passed
BUG=1399648
TBR=edheyl
OCL=111902
Change-Id: I854b01553dd92bbf9c864f5a9bd51a3d665f0ac2
Signed-off-by: Glenn Kasten <gkasten@google.com>
commit b9f3c26032be7a6ea01a10d93d94826f449e68ab
Author: Dave Bort <dbort@google.com>
Date: Fri Jan 18 14:51:05 2008 -0800
Rename "Makefile" to "Android.mk" throughout the tree.
For <http://b/issue?id=960416>.
I've tested this as much as I can, but 1500 open files =
easy to mess things up. Please let me know if there's
a problem rather than rolling back this change.
PRESUBMIT=passed
BUG=960416
TBR=joeo
OCL=46537
Change-Id: I5a404caf0f398a7afa7ae7abaf2f2a1c6ab490eb
Signed-off-by: Glenn Kasten <gkasten@google.com>
commit 0c22a9a44c4103483fba1d944acf1354c5eb1617
Author: Mathias Agopian <mathias@google.com>
Date: Mon Oct 29 23:44:25 2007 -0700
Tweak the SINC resampler parameters and double the performance. It's using about 10% CPU in the worse case now.
Change-Id: I50ac7e6c6702a427fa36ab6d976c507155057507
Signed-off-by: Glenn Kasten <gkasten@google.com>
commit b85e41487983ad085b859acf8251e7e54480308a
Author: Mathias Agopian <mathias@google.com>
Date: Mon Oct 29 04:34:36 2007 -0700
A sinc resampler for Audioflinger. It's not enabled yet, but fully functional and apparently working. It need more "quality" tests. In the 48->44 KHz, it takes about 25% of the CPU time.
Change-Id: I80eb5185e13ebdb907e0b85c49ba1272c23d60ec
Signed-off-by: Glenn Kasten <gkasten@google.com>
commit ba3949ef17cac2ba71cc3096c413782a49c922e5
Author: Mathias Agopian <mathias@google.com>
Date: Thu Aug 23 21:01:28 2007 -0700
fix a few small typos in the FIR computation
Change-Id: I6e56b514fe520f30f7487f85c64ea5d2a7c19b40
Signed-off-by: Glenn Kasten <gkasten@google.com>
commit 7474bfa7de2604021963794dddfe44985648db6a
Author: Mathias Agopian <mathias@google.com>
Date: Thu Aug 23 03:16:02 2007 -0700
This is a tool to compute the the reconstruction filter coefficients for a sinc audio resampler.
Change-Id: I99be2505139b8e0e7647200e1647509d4f7e6067
Signed-off-by: Glenn Kasten <gkasten@google.com>
Bug: 7577965
Change-Id: I2c84a9283a1668723bad83e1a119c849c88c3e6b
ndroid.mk
udioResampler.cpp
udioResamplerSinc.cpp
udioResamplerSinc.h
udio-resampler/AudioResamplerCoefficients.cpp
udio-resampler/dnsampler_filter_coefficients_x128_10112011.h
udio-resampler/filter_coefficients.h
udio-resampler/resampler_filter_coefficients_10042011.h
est-resample.cpp
|
| da398117aff410e1856299450ef6ca4faddad75c |
20-Nov-2012 |
Eric Laurent <elaurent@google.com> |
Merge "AudioFlinger files reorganization"
|
| eac37593f82855e980524f8c8c4441c21bba18e6 |
20-Nov-2012 |
Eric Laurent <elaurent@google.com> |
Merge "AudioFlinger files reorganization"
|
| ca7cc8273ffd88b9b89655808ee7e3df74162b83 |
19-Nov-2012 |
Eric Laurent <elaurent@google.com> |
AudioFlinger files reorganization
Audioflinger.cpp and Audioflinger.h files must be split to
improve readability and maintainability.
This CL splits the files as follows:
AudioFlinger.cpp split into:
- AudioFlinger.cpp: implementation of IAudioflinger interface and global methods
- AFThreads.cpp: implementation of ThreadBase, PlaybackThread, MixerThread,
DuplicatingThread, DirectOutputThread and RecordThread.
- AFTracks.cpp: implementation of TrackBase, Track, TimedTrack, OutputTrack,
RecordTrack, TrackHandle and RecordHandle.
- AFEffects.cpp: implementation of EffectModule, EffectChain and EffectHandle.
AudioFlinger.h is modified by inline inclusion of header files containing
the declaration of complex inner classes:
- AFThreads.h: ThreadBase, PlaybackThread, MixerThread, DuplicatingThread,
DirectOutputThread and RecordThread
- AFEffects.h: EffectModule, EffectChain and EffectHandle
AFThreads.h includes the follownig headers inline:
- AFTrackBase.h: TrackBase
- AFPlaybackTracks: Track, TimedTrack, OutputTrack
- AFRecordTracks: RecordTrack
Change-Id: I512ebc3a51813ab7a4afccc9a538b18125165c4c
ndroid.mk
udioFlinger.cpp
udioFlinger.h
ffects.cpp
ffects.h
laybackTracks.h
ecordTracks.h
hreads.cpp
hreads.h
rackBase.h
racks.cpp
|
| 81784c37c61b09289654b979567a42bf73cd2b12 |
19-Nov-2012 |
Eric Laurent <elaurent@google.com> |
AudioFlinger files reorganization
Audioflinger.cpp and Audioflinger.h files must be split to
improve readability and maintainability.
This CL splits the files as follows:
AudioFlinger.cpp split into:
- AudioFlinger.cpp: implementation of IAudioflinger interface and global methods
- AFThreads.cpp: implementation of ThreadBase, PlaybackThread, MixerThread,
DuplicatingThread, DirectOutputThread and RecordThread.
- AFTracks.cpp: implementation of TrackBase, Track, TimedTrack, OutputTrack,
RecordTrack, TrackHandle and RecordHandle.
- AFEffects.cpp: implementation of EffectModule, EffectChain and EffectHandle.
AudioFlinger.h is modified by inline inclusion of header files containing
the declaration of complex inner classes:
- AFThreads.h: ThreadBase, PlaybackThread, MixerThread, DuplicatingThread,
DirectOutputThread and RecordThread
- AFEffects.h: EffectModule, EffectChain and EffectHandle
AFThreads.h includes the follownig headers inline:
- AFTrackBase.h: TrackBase
- AFPlaybackTracks: Track, TimedTrack, OutputTrack
- AFRecordTracks: RecordTrack
Change-Id: I512ebc3a51813ab7a4afccc9a538b18125165c4c
ndroid.mk
udioFlinger.cpp
udioFlinger.h
ffects.cpp
ffects.h
laybackTracks.h
ecordTracks.h
hreads.cpp
hreads.h
rackBase.h
racks.cpp
|
| bc9ea63daaad764d242ce0eadfeb697abb9ce34e |
20-Nov-2012 |
Mathias Agopian <mathias@google.com> |
fix a typo in SINC resampler that prevented tracks to be mixed
we were always erasing the current mix instead of mixing into it.
Change-Id: Ib229245f9e5a0d384f1727640a59e9f0469211a2
udioResamplerSinc.cpp
|
| d6864faccd19096b72d32481a3cc475e6d36e50d |
20-Nov-2012 |
Mathias Agopian <mathias@google.com> |
fix a typo in SINC resampler that prevented tracks to be mixed
we were always erasing the current mix instead of mixing into it.
Change-Id: Ib229245f9e5a0d384f1727640a59e9f0469211a2
udioResamplerSinc.cpp
|
| c45128af9231a6e26c4deef798b79e74e4fad904 |
14-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Clean up channel count and channel mask
Channel count is uint32_t.
Remove redundant mask parameter to AudioTrack::createTrack_l()
and AudioRecord::openRecord_l().
Change-Id: I5dc2b18eb609b2c0dc3091994cbaa4628062c17f
udioFlinger.cpp
udioFlinger.h
|
| a42ff007a17d63df22c60dd5e5fd811ee45ca1b3 |
14-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Clean up channel count and channel mask
Channel count is uint32_t.
Remove redundant mask parameter to AudioTrack::createTrack_l()
and AudioRecord::openRecord_l().
Change-Id: I5dc2b18eb609b2c0dc3091994cbaa4628062c17f
udioFlinger.cpp
udioFlinger.h
|
| 7aeff3f2197de81271c3e5547486f3407be56182 |
17-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Merge ""if" statements use curly braces per media style"
|
| aa25c0321bd4b9eb46a4d1cb4ac5fa47af30eeb4 |
17-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Merge ""if" statements use curly braces per media style"
|
| 645f38376b042759ee3cfe78b7e97a6867b65814 |
17-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix time vs. bytes units bug in getRenderPosition"
|
| 77035d10a740914313500811b31a90ab948bd267 |
17-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix time vs. bytes units bug in getRenderPosition"
|
| 0dec3d6af5ad8783149134e594aadbe5dc339b4d |
17-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Don't use control block frame count after create"
|
| 67ba2b6b8018ce7556cdd880dbf28d06a92ebb78 |
17-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Don't use control block frame count after create"
|
| d7101432aa28f18b1510d9c186a27eecbeba46b2 |
14-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Don't use control block frame count after create
This is part of a series to clean up the control block.
Change-Id: I7f4cb05aef63053f8e2ab05b286d302260ef4758
udioFlinger.cpp
udioFlinger.h
|
| b603744e96b07b1d5bf745bde593fb2c025cefcf |
14-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Don't use control block frame count after create
This is part of a series to clean up the control block.
Change-Id: I7f4cb05aef63053f8e2ab05b286d302260ef4758
udioFlinger.cpp
udioFlinger.h
|
| a204922b4c04633a4e9896db3c40763bcf921ade |
23-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
"if" statements use curly braces per media style
Change-Id: I130e7849fd1da7a0b7fe56c3c53919d26e3843b8
udioFlinger.cpp
|
| d65d73c4ae74d084751b417615a78cbe7a51372a |
23-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
"if" statements use curly braces per media style
Change-Id: I130e7849fd1da7a0b7fe56c3c53919d26e3843b8
udioFlinger.cpp
|
| 411e447c4b90298f5ff635429c53f94fbce4fff9 |
02-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Don't explicitly log tid
If needed, it can be obtained with adb logcat -v threadtime
Change-Id: I91b3911d20f7bcfc3361db4052db21ff9181f1cf
udioFlinger.cpp
udioPolicyService.cpp
|
| 827e5f1237757aee78b677efcf0f7c44fd0dd3d8 |
02-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Don't explicitly log tid
If needed, it can be obtained with adb logcat -v threadtime
Change-Id: I91b3911d20f7bcfc3361db4052db21ff9181f1cf
udioFlinger.cpp
udioPolicyService.cpp
|
| b59a5026ecdb3dc283c448e5bbd51b1c6d42b946 |
16-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Fix time vs. bytes units bug in getRenderPosition
Rename correctLatency since it requires thread to be locked.
Use size_t for byte and frame counts.
Change-Id: I178fdd18bdb823813b9563927bdff8c0d28ca5a5
udioFlinger.cpp
udioFlinger.h
|
| 26c77556efc30800466b60b3975bc35a70c8c28b |
16-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Fix time vs. bytes units bug in getRenderPosition
Rename correctLatency since it requires thread to be locked.
Use size_t for byte and frame counts.
Change-Id: I178fdd18bdb823813b9563927bdff8c0d28ca5a5
udioFlinger.cpp
udioFlinger.h
|
| d983364b3655a547b55bb11dbe148103198c011d |
15-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Static AudioTrack plays twice initially
Bug: 7528721
Change-Id: I10bc16a26f33dba6572b730a170cb3bf00e68e30
udioFlinger.cpp
|
| 35436ad21902edf83e3c1307b45b38d13137d31c |
15-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Static AudioTrack plays twice initially
Bug: 7528721
Change-Id: I10bc16a26f33dba6572b730a170cb3bf00e68e30
udioFlinger.cpp
|
| 56938ef8dc82b49ddbc201a8855c91ba46d73bfb |
16-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use size_t for frame counts"
|
| aac817e6685a51843e84c23d970b3d559e74e8f6 |
16-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use size_t for frame counts"
|
| 7da35f231725aa6e0b2ac8714a7d66471368df77 |
14-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Use size_t for frame counts
Also fix typo: bufferCount should be frameCount.
Change-Id: Ibed539504db75ef99dc21c8ff1bf2987122063a5
udioFlinger.cpp
udioFlinger.h
|
| e33054eb968cbf8ccaee1b0ff0301403902deed6 |
14-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Use size_t for frame counts
Also fix typo: bufferCount should be frameCount.
Change-Id: Ibed539504db75ef99dc21c8ff1bf2987122063a5
udioFlinger.cpp
udioFlinger.h
|
| 5b78fc92d9e64623be66a5c3031b54dcd8ed28f9 |
15-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Static AudioTrack plays twice initially
Bug: 7528721
Change-Id: I10bc16a26f33dba6572b730a170cb3bf00e68e30
udioFlinger.cpp
|
| 4bd7e5436f9c308503d72e80804fb5637fda1584 |
15-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Static AudioTrack plays twice initially
Bug: 7528721
Change-Id: I10bc16a26f33dba6572b730a170cb3bf00e68e30
udioFlinger.cpp
|
| 520a9af9438c29b24e328dd2b7a287c7a96a4e6b |
21-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Clean up frame size in AudioTrack and AudioFlinger
TrackBase::mFrameSize, mChannelMask, and mChannelCount are now const.
Use TrackBase::mFrameSize instead of re-calculating frame size.
AudioFlinger only sees 16-bit PCM format, conversion from 8-bit is
now entirely on the client side. Previously a small part of the
responsibility was on server side also.
size_t is unsigned, so use %u in logs.
Fix theoretical bug where TrackBase constructor was over-allocating space
for non-linear AudioTrack or 8-bit PCM AudioRecord (probably benign).
Change-Id: I7cbbba0bf4dba29ea751d8af341ab8e5cbbdc206
udioFlinger.cpp
udioFlinger.h
|
| 60a839204713e0f8258d082af83262b1eb33a6c3 |
21-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Clean up frame size in AudioTrack and AudioFlinger
TrackBase::mFrameSize, mChannelMask, and mChannelCount are now const.
Use TrackBase::mFrameSize instead of re-calculating frame size.
AudioFlinger only sees 16-bit PCM format, conversion from 8-bit is
now entirely on the client side. Previously a small part of the
responsibility was on server side also.
size_t is unsigned, so use %u in logs.
Fix theoretical bug where TrackBase constructor was over-allocating space
for non-linear AudioTrack or 8-bit PCM AudioRecord (probably benign).
Change-Id: I7cbbba0bf4dba29ea751d8af341ab8e5cbbdc206
udioFlinger.cpp
udioFlinger.h
|
| 1127d65d536ebbe447ee17ce0926a7ce4a2a3c08 |
14-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Use uint32_t for sample rate
Change-Id: Ie240b48fb54b08359f69ecd4e5f8bda3d15cbe80
udioFlinger.cpp
udioFlinger.h
|
| 3b16c766d1ae2cfd8487e8ffb2b23936fc0a8e17 |
14-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Use uint32_t for sample rate
Change-Id: Ie240b48fb54b08359f69ecd4e5f8bda3d15cbe80
udioFlinger.cpp
udioFlinger.h
|
| 1513ad2d2de0962cc3b3121e6fae73d8ee1a4639 |
15-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Update audio comments"
|
| b4a17e834b718eff1ba2eac4232de6e73a4bf9f5 |
15-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Update audio comments"
|
| a9749e3468fda2288dda65d1ef52e0997f6d8897 |
15-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix build warnings"
|
| acc8706f91f799ea63c10ecad4959c55e0873d93 |
15-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix build warnings"
|
| 656130a7bcff790dc9311754189d46bd4b911fbf |
14-Nov-2012 |
Mathias Agopian <mathias@google.com> |
Merge changes I2ce3479d,Ibb56664d
* changes:
more optimizations...
refactor code to improve neon code
|
| 30ca3be314993fb4155bee1a17a4b76354cd6ef9 |
14-Nov-2012 |
Mathias Agopian <mathias@google.com> |
Merge changes I2ce3479d,Ibb56664d
* changes:
more optimizations...
refactor code to improve neon code
|
| 020f79fd967736d36c7ec860814e2e93866a020f |
07-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Update audio comments
Change-Id: I85d7d2f6381b251db5695202fec75128883a8662
udioFlinger.h
tateQueue.h
|
| 22eb4e239fbe9103568147d566d7482e480350b8 |
07-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Update audio comments
Change-Id: I85d7d2f6381b251db5695202fec75128883a8662
udioFlinger.h
tateQueue.h
|
| 98d9922247ef21b0faf8c7d817650b8327052a4a |
14-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Fix build warnings
Change-Id: Ic43bcca166a529a6431711b05a7fa21849b6a38b
est-resample.cpp
|
| b26e3e9f2ab0334bff21a4fa4851dbf6e57fba5d |
14-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Fix build warnings
Change-Id: Ic43bcca166a529a6431711b05a7fa21849b6a38b
est-resample.cpp
|
| e2a0586eae45a3eb1fad0f28ca2f8b0204d78df6 |
13-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Rename TrackBase::mFrameCount to mStepCount
This prepares for adding a new field TrackBase::mFrameCount
with a different meaning.
Change-Id: I6bbe2c59f2a882be57caeec2e2e06f439a0e9e83
udioFlinger.cpp
udioFlinger.h
|
| 9f2016d9adfb4f88fa0bbfcfa5954f79160db595 |
13-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Rename TrackBase::mFrameCount to mStepCount
This prepares for adding a new field TrackBase::mFrameCount
with a different meaning.
Change-Id: I6bbe2c59f2a882be57caeec2e2e06f439a0e9e83
udioFlinger.cpp
udioFlinger.h
|
| 5aab59a2bd0a2cd80240ffd66c1b963b5fe06d65 |
12-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Move frame size out of the control block
This is part of a series to clean up the control block.
Change-Id: Ifab1c42ac0f8be704e571b292713cd2250d12a3f
udioFlinger.cpp
udioFlinger.h
|
| 83a0382dc17364567667a4e6135db43f5bd92efc |
12-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Move frame size out of the control block
This is part of a series to clean up the control block.
Change-Id: Ifab1c42ac0f8be704e571b292713cd2250d12a3f
udioFlinger.cpp
udioFlinger.h
|
| 5d3370d0db4dc97fa413a302e426fb24e87462be |
10-Nov-2012 |
Mathias Agopian <mathias@google.com> |
more optimizations...
calculate the offsets from the phase differently, this happens
to reduce the register pressure in the main loop, which in turns
allows the compiler to generate much better code (doesn't need
to spill a lot of stuff on the stack).
this gives another 15% performance increase
Change-Id: I2ce3479dd48b9e6941adb80e6d443d6e14d64d96
udioResamplerSinc.cpp
udioResamplerSinc.h
|
| 7492a7ff46a75b5d8e10ae11d4ad50429cf945ce |
10-Nov-2012 |
Mathias Agopian <mathias@google.com> |
more optimizations...
calculate the offsets from the phase differently, this happens
to reduce the register pressure in the main loop, which in turns
allows the compiler to generate much better code (doesn't need
to spill a lot of stuff on the stack).
this gives another 15% performance increase
Change-Id: I2ce3479dd48b9e6941adb80e6d443d6e14d64d96
udioResamplerSinc.cpp
udioResamplerSinc.h
|
| 3c11ff2f409a0abb5b9d8ffd5e13cc42cda67fdc |
10-Nov-2012 |
Mathias Agopian <mathias@google.com> |
refactor code to improve neon code
we want to make sure we don't transfer data from the
neon unit to the arm register file, as this can be quite
slow. instead we do all the calculation on the neon side
and write the result directly to main memory.
Change-Id: Ibb56664d3ab03098ae2798b75e2b6927ac900187
udioResamplerSinc.cpp
udioResamplerSinc.h
|
| 0d585c85524eb5d398fadff5ca8dd43939ed9cb4 |
10-Nov-2012 |
Mathias Agopian <mathias@google.com> |
refactor code to improve neon code
we want to make sure we don't transfer data from the
neon unit to the arm register file, as this can be quite
slow. instead we do all the calculation on the neon side
and write the result directly to main memory.
Change-Id: Ibb56664d3ab03098ae2798b75e2b6927ac900187
udioResamplerSinc.cpp
udioResamplerSinc.h
|
| 3f6f4393d0adf4465959059d69798aed55e34876 |
09-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Move buffers pointer out of the control block"
|
| e35a582d02586cb43ad9e344138b4edad62868b3 |
09-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Move buffers pointer out of the control block"
|
| c6ccbabd2800e749bc86a34d2ae233f99102ba51 |
05-Nov-2012 |
Mathias Agopian <mathias@google.com> |
NEON optimized SINC resampler
this currently gives us a 60% to 80% boost depending
on the quality level selected.
Change-Id: I7db385007e811ed7bffe5fd3403b44e300894f5b
udioResamplerSinc.cpp
est-resample.cpp
|
| ad9af03c4b491912239fc8c97a3ad0d342a33303 |
05-Nov-2012 |
Mathias Agopian <mathias@google.com> |
NEON optimized SINC resampler
this currently gives us a 60% to 80% boost depending
on the quality level selected.
Change-Id: I7db385007e811ed7bffe5fd3403b44e300894f5b
udioResamplerSinc.cpp
est-resample.cpp
|
| af03dfb73d05dfabdf55640aff8b2256fdd78663 |
05-Nov-2012 |
Mathias Agopian <mathias@google.com> |
minor cleanups
Change-Id: Ia12ee4fb59e90221761bec85e6450db29197591f
udioResamplerSinc.cpp
|
| 7aa7ed773040ea60bbe0a2a6ea949d62802304a4 |
05-Nov-2012 |
Mathias Agopian <mathias@google.com> |
minor cleanups
Change-Id: Ia12ee4fb59e90221761bec85e6450db29197591f
udioResamplerSinc.cpp
|
| 2f6226ad7a3fa81dc102001fbc1d1288e55a0947 |
08-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Move buffers pointer out of the control block
This is part of a series to clean up the control block.
Change-Id: Ie474557db7cb360f2d9a0f11600a68f5a3d46f07
udioFlinger.cpp
udioFlinger.h
|
| b929e417853694e37aba1ef4399f188987b709d9 |
08-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Move buffers pointer out of the control block
This is part of a series to clean up the control block.
Change-Id: Ie474557db7cb360f2d9a0f11600a68f5a3d46f07
udioFlinger.cpp
udioFlinger.h
|
| ba85098eb31bd2637db49816f0591361211024f2 |
07-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Remove CBLK_DIRECTION from control block flags
This is part of a series to clean up the control block.
Change-Id: I0265fece3247356b585d4d48fbda6f37aea8a851
udioFlinger.cpp
udioFlinger.h
|
| 864585df53eb97c31e77b3ad7c0d89e4f9b42588 |
07-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Remove CBLK_DIRECTION from control block flags
This is part of a series to clean up the control block.
Change-Id: I0265fece3247356b585d4d48fbda6f37aea8a851
udioFlinger.cpp
udioFlinger.h
|
| 891b11da870ad3f860c1d2610ef4d8836ed6c590 |
07-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove CBLK_FAST from control block flags"
|
| 8fa3ad0e4c77f7868348ad2a4cacc34c34e7c39e |
07-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove CBLK_FAST from control block flags"
|
| c26741598a840f7c1d12ee457fb26f68fcdbcc70 |
07-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Remove CBLK_FAST from control block flags
This is part of a series to clean up the control block.
Change-Id: Ic881a3560d9547cb63fcc0cefec87aa3da480e0d
udioFlinger.cpp
udioFlinger.h
|
| e0b07179a48ee50fda931d2aa1b3c751d167e4d7 |
07-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Remove CBLK_FAST from control block flags
This is part of a series to clean up the control block.
Change-Id: Ic881a3560d9547cb63fcc0cefec87aa3da480e0d
udioFlinger.cpp
udioFlinger.h
|
| 3929402c7e3d9865cdeabedf291e1c4a1db48ce7 |
07-Nov-2012 |
Mathias Agopian <mathias@google.com> |
Merge changes I5d62a6c2,Iebf76958,I8068f0f7
* changes:
improve resample test
change how we store the FIR coefficients
improve SINC resampler performance
|
| 879256758603bac0e4718ab88db145f19f70027b |
07-Nov-2012 |
Mathias Agopian <mathias@google.com> |
Merge changes I5d62a6c2,Iebf76958,I8068f0f7
* changes:
improve resample test
change how we store the FIR coefficients
improve SINC resampler performance
|
| ac65726b26a63635d3461f96b4b979add5a8f7dc |
05-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Merge duplicate code in TrackBase::TrackBase()
Change-Id: Id8a7db4d94888796a9dcbed4cac99941705cd174
udioFlinger.cpp
|
| 287fedb1b1430f138c1f583869b10294773945be |
05-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Merge duplicate code in TrackBase::TrackBase()
Change-Id: Id8a7db4d94888796a9dcbed4cac99941705cd174
udioFlinger.cpp
|
| d12b033552585cd8761ba6241f21e4cea4a21878 |
05-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Simplify control block flag names
Use only one symbol per flag
Change-Id: Ia3582e2134abd60c896d11337face65383e79c7c
udioFlinger.cpp
|
| 9c5fdd83f9b9f49be35107971feb33528d60b945 |
05-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Simplify control block flag names
Use only one symbol per flag
Change-Id: Ia3582e2134abd60c896d11337face65383e79c7c
udioFlinger.cpp
|
| 7df17073f1783421d6ed2013e6b55c60283d1474 |
05-Nov-2012 |
Mathias Agopian <mathias@google.com> |
improve resample test
- handle stereo input
- input file can now be ommited, in this case
a linear chirp will be used automatically
- better usage information
Change-Id: I5d62a6c26a9054a1c1a517a065b4df5a2cdcda22
est-resample.cpp
|
| 3f71761cab8a08e4ae9e4cf8cb8f1b82643825b2 |
05-Nov-2012 |
Mathias Agopian <mathias@google.com> |
improve resample test
- handle stereo input
- input file can now be ommited, in this case
a linear chirp will be used automatically
- better usage information
Change-Id: I5d62a6c26a9054a1c1a517a065b4df5a2cdcda22
est-resample.cpp
|
| e8299af5665bf9c396466c33b9b16a84fe78c7f6 |
04-Nov-2012 |
Mathias Agopian <mathias@google.com> |
change how we store the FIR coefficients
The coefficient table is now transposed and shows
much better its polyphase nature: we now have a FIR
per line, each line corresponding to a phase.
This doesn't change at all the results produced by
the filter, but allows us to make slightly better
use of the data cache and improves performance a bit
(although not as much as I thought it would).
The main benefit is that it is the first step
before we can make much larger optimizations
(like using NEON).
Change-Id: Iebf7695825dcbd41f25861efcaefbaa3365ecb43
udioResamplerSinc.cpp
udioResamplerSinc.h
udio-resampler/filter_coefficients.h
|
| 46afbec3743f1d799f185273ff897d1f8e0175dd |
04-Nov-2012 |
Mathias Agopian <mathias@google.com> |
change how we store the FIR coefficients
The coefficient table is now transposed and shows
much better its polyphase nature: we now have a FIR
per line, each line corresponding to a phase.
This doesn't change at all the results produced by
the filter, but allows us to make slightly better
use of the data cache and improves performance a bit
(although not as much as I thought it would).
The main benefit is that it is the first step
before we can make much larger optimizations
(like using NEON).
Change-Id: Iebf7695825dcbd41f25861efcaefbaa3365ecb43
udioResamplerSinc.cpp
udioResamplerSinc.h
udio-resampler/filter_coefficients.h
|
| 50ebdf2086b645b9b703a6d489238767a9afb34f |
04-Nov-2012 |
Mathias Agopian <mathias@google.com> |
improve SINC resampler performance
The improvement is about 60% by just tweaking a few
things to help the compiler generate better code.
It turns out that inlining too much stuff manually was hurting us.
Change-Id: I8068f0f75051f95ac600e50ce552572dd1e8c304
udioResamplerSinc.cpp
udioResamplerSinc.h
|
| a798c97386a842d06d290797ba5dce95d031332a |
04-Nov-2012 |
Mathias Agopian <mathias@google.com> |
improve SINC resampler performance
The improvement is about 60% by just tweaking a few
things to help the compiler generate better code.
It turns out that inlining too much stuff manually was hurting us.
Change-Id: I8068f0f75051f95ac600e50ce552572dd1e8c304
udioResamplerSinc.cpp
udioResamplerSinc.h
|
| fb242fa66643d1bd5362077a44c02c86c8064564 |
02-Nov-2012 |
Mathias Agopian <mathias@google.com> |
Merge "new coefficients for the vhq resampler"
|
| f27bdebd0c611f9618577f65346f44574146dc00 |
02-Nov-2012 |
Mathias Agopian <mathias@google.com> |
Merge "new coefficients for the vhq resampler"
|
| c5b0b9a13a29be483173f092663ed17ea306ac0d |
02-Nov-2012 |
Mathias Agopian <mathias@google.com> |
new coefficients for the vhq resampler
previous coefficients were provided by a 3rd party and didn't have a
way to re-generate them. we're now using the 'fir' utility.
the performance of the filter is virtually identical, except for
the down-sampling case which seems slightly better now:
It looks like both the previous and new coefficients are generating
some sort of clipping for full-scale signals in the down-sampling case
(although the new ones seem better), the reason for that is
unknown (see bug: 7453062)
Also updated the HQ coefficients for the down-samplers, previous ones
were a little bit too conservative -- the new ones push the cut-off
frequency up by about 1 KHz.
Change-Id: I54a827b5c707c7cc41268ed01283758dce1d7647
udioResamplerSinc.cpp
udio-resampler/AudioResamplerCoefficients.cpp
udio-resampler/dnsampler_filter_coefficients_x128_10112011.h
udio-resampler/filter_coefficients.h
udio-resampler/resampler_filter_coefficients_10042011.h
|
| 4ed475d3ad4231370371e14a94779c5d300eb3c5 |
02-Nov-2012 |
Mathias Agopian <mathias@google.com> |
new coefficients for the vhq resampler
previous coefficients were provided by a 3rd party and didn't have a
way to re-generate them. we're now using the 'fir' utility.
the performance of the filter is virtually identical, except for
the down-sampling case which seems slightly better now:
It looks like both the previous and new coefficients are generating
some sort of clipping for full-scale signals in the down-sampling case
(although the new ones seem better), the reason for that is
unknown (see bug: 7453062)
Also updated the HQ coefficients for the down-samplers, previous ones
were a little bit too conservative -- the new ones push the cut-off
frequency up by about 1 KHz.
Change-Id: I54a827b5c707c7cc41268ed01283758dce1d7647
udioResamplerSinc.cpp
udio-resampler/AudioResamplerCoefficients.cpp
udio-resampler/dnsampler_filter_coefficients_x128_10112011.h
udio-resampler/filter_coefficients.h
udio-resampler/resampler_filter_coefficients_10042011.h
|
| 8af901cdea0af7e536579dee6d56e69987035a01 |
01-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Line length 100
Change-Id: Ib28fd7b9ce951a6933f006e7f8812ba617625530
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
udioMixer.h
udioPolicyService.cpp
udioPolicyService.h
est-resample.cpp
|
| 85ab62c4b433df3f1a9826bed1c9bec07a86c750 |
01-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Line length 100
Change-Id: Ib28fd7b9ce951a6933f006e7f8812ba617625530
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
udioMixer.h
udioPolicyService.cpp
udioPolicyService.h
est-resample.cpp
|
| 8c32734c1e2dda852011fc46d0caded971464bc2 |
30-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
Save copy of mic input, disabled by default
Change-Id: I4f5e95a5ddf016530d1b2747a0a5ca0962caabda
ndroid.mk
udioFlinger.cpp
udioFlinger.h
|
| d06785bebf7e43d4a011b62a252771373ada910c |
30-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
Save copy of mic input, disabled by default
Change-Id: I4f5e95a5ddf016530d1b2747a0a5ca0962caabda
ndroid.mk
udioFlinger.cpp
udioFlinger.h
|
| 901475744e15d8a4728cb111e510624377abaec3 |
31-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "StateQueue comments"
|
| a2757a0ef4ccffd01864dcc643726d26838d562b |
31-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "StateQueue comments"
|
| be58bd19f4b99e04651cff0dfc968248f5b0e440 |
31-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove obsolete references to libmedia_native"
|
| b7d55a839356ad55316b5dd5b83396b69d391ce3 |
31-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove obsolete references to libmedia_native"
|
| d55d23209bb90ff7bd7443d1b54fa07c0ed8d14f |
30-Oct-2012 |
Mathias Agopian <mathias@google.com> |
Merge changes Ide218785,I86d0bb82
* changes:
fix SINC resampler on non ARM architectures
fix another issue with generating FIR coefficients
|
| 670d4b7520bca8c6b3c59db2640698358d83aca6 |
30-Oct-2012 |
Mathias Agopian <mathias@google.com> |
Merge changes Ide218785,I86d0bb82
* changes:
fix SINC resampler on non ARM architectures
fix another issue with generating FIR coefficients
|
| d7a3c142b3ce4ed26e025533f57bef9e4ec58740 |
30-Oct-2012 |
Mathias Agopian <mathias@google.com> |
fix SINC resampler on non ARM architectures
make sure the C version of the code generates the same
output than the ARM assemply version.
Change-Id: Ide218785c35d02598b2d7278e646b1b178148698
udioResamplerSinc.cpp
|
| 1f09b4ada212d259b531228db67ca160d280275c |
30-Oct-2012 |
Mathias Agopian <mathias@google.com> |
fix SINC resampler on non ARM architectures
make sure the C version of the code generates the same
output than the ARM assemply version.
Change-Id: Ide218785c35d02598b2d7278e646b1b178148698
udioResamplerSinc.cpp
|
| cca56d5784c02a5f02ce5e47c3d6583d71a2469b |
30-Oct-2012 |
Mathias Agopian <mathias@google.com> |
fix another issue with generating FIR coefficients
the impulse response of a low-pass is 2*f*sinc(2*pi*f*k), we were
missing the 2*f scale factor. This explains why we were seeing
clipping and had to manually scale the filter down.
Change-Id: I86d0bb82ecdd99681c8ba5a8112a8257bf6f0186
udioResamplerSinc.cpp
|
| d88a051aff15fdf5c57e1e5a4083bbd9635af3ad |
30-Oct-2012 |
Mathias Agopian <mathias@google.com> |
fix another issue with generating FIR coefficients
the impulse response of a low-pass is 2*f*sinc(2*pi*f*k), we were
missing the 2*f scale factor. This explains why we were seeing
clipping and had to manually scale the filter down.
Change-Id: I86d0bb82ecdd99681c8ba5a8112a8257bf6f0186
udioResamplerSinc.cpp
|
| faa4420344216ccbad15a49aa7c1cd6141532fcb |
30-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Remove obsolete references to libmedia_native
Bug: 6654403
Change-Id: I3993d62987cf0dd85db10bf002a5cce53d4f01bd
ndroid.mk
|
| c41590251aa84c078c942d258e838aad814b73a5 |
30-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Remove obsolete references to libmedia_native
Bug: 6654403
Change-Id: I3993d62987cf0dd85db10bf002a5cce53d4f01bd
ndroid.mk
|
| b387e0f0e3338b4e19055d6ec2b79c04314251b3 |
30-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Document AudioMixer hard-coded limits"
|
| 0eac0714e9fbd6e43b1fd13a38131800b2a81377 |
30-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Document AudioMixer hard-coded limits"
|
| f84d5761c5f12bb75af492de27d9bc5491e5667d |
30-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use ALOGVV instead of commented ALOGV"
|
| db3a20ea484514f7f23d4ef3c30fe4780ab940d2 |
30-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use ALOGVV instead of commented ALOGV"
|
| 3b0389dd5f611db1f65dcae88d0f6b332e447d7a |
30-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Clean up constructor and derivation whitespace"
|
| bf1d047d6759c624139bfe9897dc3062d2e446e2 |
30-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Clean up constructor and derivation whitespace"
|
| a6d7d340cee564d41e7594a22233cab3d1ee26d9 |
30-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Mark volume fields private"
|
| eb79904784c9479ae6fb60202c47e5287c058c4a |
30-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Mark volume fields private"
|
| 7c4349fe6e1e4faaa0c0db40a74f2ccbac46f57e |
30-Oct-2012 |
Mathias Agopian <mathias@google.com> |
fir a typo that caused up-sampling coefficiens to be wrong
up-sample coefficient were generated with a cut-off frequency of 24KHz
intead of ~20KHz, which caused more aliasing in the audible band.
also increased the attenuation to 1.3 dB on both up and down
sampling coefficient to avoid clipping.
Change-Id: Ie8aeecf1429190541b656810c6716b6aae5ece2e
udioResamplerSinc.cpp
|
| b4b75b47c2a4248e60bbc3229d6acc4d5f872431 |
30-Oct-2012 |
Mathias Agopian <mathias@google.com> |
fir a typo that caused up-sampling coefficiens to be wrong
up-sample coefficient were generated with a cut-off frequency of 24KHz
intead of ~20KHz, which caused more aliasing in the audible band.
also increased the attenuation to 1.3 dB on both up and down
sampling coefficient to avoid clipping.
Change-Id: Ie8aeecf1429190541b656810c6716b6aae5ece2e
udioResamplerSinc.cpp
|
| 3e48fd11e3fca5c1d03354a99657298e110be56a |
30-Oct-2012 |
Mathias Agopian <mathias@google.com> |
test-resample: clip instead of overflowing
Change-Id: I550e5a59e51c11e1095ca338222b094f92b96878
est-resample.cpp
|
| 9aec8c3f7f72cd36a8e3d7aafc1149f50514087a |
30-Oct-2012 |
Mathias Agopian <mathias@google.com> |
test-resample: clip instead of overflowing
Change-Id: I550e5a59e51c11e1095ca338222b094f92b96878
est-resample.cpp
|
| cbe4b36e67baba4ec1a4d7a9db9461e09c714f0c |
19-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Use ALOGVV instead of commented ALOGV
Change-Id: I17bdb5274877760c7edaa31416020d349082e593
udioFlinger.cpp
|
| 26dd66e8ea7a1abf28c33196fc77822f68a718af |
19-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Use ALOGVV instead of commented ALOGV
Change-Id: I17bdb5274877760c7edaa31416020d349082e593
udioFlinger.cpp
|
| 67360781c2d139a71d924ae24880b56adb7c1605 |
27-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
StateQueue comments
Change-Id: Id8c76d1e2ab5201e7ab3875f1dbcc0a126e611a4
tateQueue.h
|
| 2188bc912a56d9bc577fcec7bf2208f49455e744 |
27-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
StateQueue comments
Change-Id: Id8c76d1e2ab5201e7ab3875f1dbcc0a126e611a4
tateQueue.h
|
| d82c750ca261d01caa405f555eef34b7af8d9b63 |
08-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Document AudioMixer hard-coded limits
Change-Id: I83ea8bed375f251260945db788bdb5f280dba12d
udioMixer.cpp
udioMixer.h
|
| 599fabc596687efa4b71b8f3ebbb957c7cae0c72 |
08-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Document AudioMixer hard-coded limits
Change-Id: I83ea8bed375f251260945db788bdb5f280dba12d
udioMixer.cpp
udioMixer.h
|
| 8a831724c52164a62e22e8e4d5f5180501e3c307 |
10-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Clean up constructor and derivation whitespace
Change-Id: I47d688a9c10c4c3c868accc34102fb402ebcac62
udioFlinger.h
|
| 274c02ee1464d8948913ac70e64e8dbb80f82ad7 |
10-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Clean up constructor and derivation whitespace
Change-Id: I47d688a9c10c4c3c868accc34102fb402ebcac62
udioFlinger.h
|
| 288a6fd72949dec167c5421d5bc977d1ccb75561 |
06-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Mark volume fields private
Change-Id: I8ffca0460195263d159aa13015c246122d8556a2
udioFlinger.h
|
| dc8a0d75bd7b1343cd65c3c7f6e0f91ca0fa6946 |
06-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Mark volume fields private
Change-Id: I8ffca0460195263d159aa13015c246122d8556a2
udioFlinger.h
|
| bcf5220780fbf24477bf8d6148581307356c97b7 |
21-Oct-2012 |
Mathias Agopian <mathias@google.com> |
a test app for the resamplers
Change-Id: I66852d90d384f1d9e77b51ad1a1ebdbaf61d0607
ndroid.mk
est-resample.cpp
|
| 0fc2cb59d5f77412f5922540d67fea81f4d1744b |
21-Oct-2012 |
Mathias Agopian <mathias@google.com> |
a test app for the resamplers
Change-Id: I66852d90d384f1d9e77b51ad1a1ebdbaf61d0607
ndroid.mk
est-resample.cpp
|
| 51ba3c7c9a4a238cdef57ad75a2d8f406ffd4515 |
26-Oct-2012 |
Mathias Agopian <mathias@google.com> |
reenable the cubic resampler
cubic resampler was disabled because it hadn't been qualified,
however after I did some tests, it does improve significantly
the sound quality over the order-1 resampler, even if it is
still quite bad.
also HIGH_QUALITY resampler was partially disabled, it's now
fully enabled. It's a big improvement over the cubic resampler
in terms of aliasing noise (it's not as good in the pass-band).
Change-Id: I70e3658c255896588642697be9eb594ff4ec0f8b
ndroid.mk
udioResampler.cpp
|
| 93d0767a8a9ee9d649eea9afac59f778e29a6a54 |
26-Oct-2012 |
Mathias Agopian <mathias@google.com> |
reenable the cubic resampler
cubic resampler was disabled because it hadn't been qualified,
however after I did some tests, it does improve significantly
the sound quality over the order-1 resampler, even if it is
still quite bad.
also HIGH_QUALITY resampler was partially disabled, it's now
fully enabled. It's a big improvement over the cubic resampler
in terms of aliasing noise (it's not as good in the pass-band).
Change-Id: I70e3658c255896588642697be9eb594ff4ec0f8b
ndroid.mk
udioResampler.cpp
|
| 3bd72cc23f74e750069a2943ad3d5c9af3be4b55 |
26-Oct-2012 |
Mathias Agopian <mathias@google.com> |
improve SINC resampler coefficients
- we increase the interpolation precision from 4 to 7 bits
this doesn't increase CPU power required, it only increases the
size of the filter table but significantly reduces the noise
introduced by the quantization of the impulse response.
- the parameters of the filter are set such that aliasing is
rejected at 80 dB below 20 KHz. Because we don't use a lot of
coefficient (to save compute power), there are quite a bit of
attenuation in the pass-band: starting at 9KHz for the
down-sampler (48 to 44.1), and starting at 13 KHz for the
up-sampler (44.1 to 48) -- the transition band is about 15 KHz.
Change-Id: I855548d2aab8a0fb0d2a2da3a364b6842d7d3838
udioResamplerSinc.cpp
udioResamplerSinc.h
|
| 443e69625d598ea578e2c838960778ce498fd773 |
26-Oct-2012 |
Mathias Agopian <mathias@google.com> |
improve SINC resampler coefficients
- we increase the interpolation precision from 4 to 7 bits
this doesn't increase CPU power required, it only increases the
size of the filter table but significantly reduces the noise
introduced by the quantization of the impulse response.
- the parameters of the filter are set such that aliasing is
rejected at 80 dB below 20 KHz. Because we don't use a lot of
coefficient (to save compute power), there are quite a bit of
attenuation in the pass-band: starting at 9KHz for the
down-sampler (48 to 44.1), and starting at 13 KHz for the
up-sampler (44.1 to 48) -- the transition band is about 15 KHz.
Change-Id: I855548d2aab8a0fb0d2a2da3a364b6842d7d3838
udioResamplerSinc.cpp
udioResamplerSinc.h
|
| d5f327e5f1d9d43208c59ae1f47187fb375fb941 |
23-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
am 3aaa4e30: am c4640c9e: Increase allowed number of VHQ resamplers to 3
* commit '3aaa4e30f7663673f3453d3f4671b1119f1802fd':
Increase allowed number of VHQ resamplers to 3
|
| 790382e5f7e18672dce955747b8afd09a5e6cf04 |
23-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
am 3aaa4e30: am c4640c9e: Increase allowed number of VHQ resamplers to 3
* commit '3aaa4e30f7663673f3453d3f4671b1119f1802fd':
Increase allowed number of VHQ resamplers to 3
|
| c4640c9eef850bb1c754bd6b477f1cc8350c6081 |
23-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Increase allowed number of VHQ resamplers to 3
Bug: 7378660
Change-Id: I69e33ca2eb4bb9bd38e2c63df62cd1130d68baf6
udioResampler.cpp
|
| f1b2a9b65a904b10f91482cc0389bea1f66a891f |
23-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Increase allowed number of VHQ resamplers to 3
Bug: 7378660
Change-Id: I69e33ca2eb4bb9bd38e2c63df62cd1130d68baf6
udioResampler.cpp
|
| 8516ae9f56fd6d2ae6c6b5534609ae2ae3725d10 |
22-Oct-2012 |
Mathias Agopian <mathias@google.com> |
am 7f4d5714: am e384d75b: Merge "Fix a typo that caused the high quality resampler to produce garbage" into jb-mr1-dev
* commit '7f4d5714f3f0ab4efaf2d0929a9f984467ef95c7':
Fix a typo that caused the high quality resampler to produce garbage
|
| df71e8e48ccf36b9f3947fb9d90cea4a5d319187 |
22-Oct-2012 |
Mathias Agopian <mathias@google.com> |
am 7f4d5714: am e384d75b: Merge "Fix a typo that caused the high quality resampler to produce garbage" into jb-mr1-dev
* commit '7f4d5714f3f0ab4efaf2d0929a9f984467ef95c7':
Fix a typo that caused the high quality resampler to produce garbage
|
| 61ea117b03b53382b5ecbc33004c7d37ea70ea8b |
21-Oct-2012 |
Mathias Agopian <mathias@google.com> |
Fix a typo that caused the high quality resampler to produce garbage
the problem is that if libaudio_resampler is present, it is those
coefficients that will always be selected, but the correct
meta-data.
Bug: 7385994
Change-Id: Ieebeb37b4dfb62a1a051bc29fae2ce056dbc6621
udioResamplerSinc.cpp
|
| ebed918320bdacdb225f679a8d7135d1acd3fe51 |
21-Oct-2012 |
Mathias Agopian <mathias@google.com> |
Fix a typo that caused the high quality resampler to produce garbage
the problem is that if libaudio_resampler is present, it is those
coefficients that will always be selected, but the correct
meta-data.
Bug: 7385994
Change-Id: Ieebeb37b4dfb62a1a051bc29fae2ce056dbc6621
udioResamplerSinc.cpp
|
| d3eb71055a81cf405ae3fac8172d3fd592ed2e4b |
19-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
am 424b4a57: am 291f824e: Remove active track when thread goes to standby
* commit '424b4a57f5e1e8bd99cb619398389a467ec41f92':
Remove active track when thread goes to standby
|
| 4687da74717842852009df4787fe041ef8c935ef |
19-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
am 424b4a57: am 291f824e: Remove active track when thread goes to standby
* commit '424b4a57f5e1e8bd99cb619398389a467ec41f92':
Remove active track when thread goes to standby
|
| 291f824e02ff517a34cfe50220b4e2b402ee998d |
19-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Remove active track when thread goes to standby
Bug: 7369232
Change-Id: I7ff9f525dad4be0aef562a53015b06ee7d3d50f1
udioFlinger.cpp
|
| 126c910f48a1e1b7a8044157d95f774d5e142316 |
19-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Remove active track when thread goes to standby
Bug: 7369232
Change-Id: I7ff9f525dad4be0aef562a53015b06ee7d3d50f1
udioFlinger.cpp
|
| ca794bc9fb43f24addff4c1915d397269d1568ab |
16-Oct-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
am aa00830e: am bbe11e46: Merge "Fix track estimation for presentation complete" into jb-mr1-dev
* commit 'aa00830e46cb7494af7217246788c1bca83b1d04':
Fix track estimation for presentation complete
|
| 426ee0f6f828b01693f83780832c167b683c004f |
16-Oct-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
am aa00830e: am bbe11e46: Merge "Fix track estimation for presentation complete" into jb-mr1-dev
* commit 'aa00830e46cb7494af7217246788c1bca83b1d04':
Fix track estimation for presentation complete
|
| a045dcafd2b77036210f5b72e79d745ad4c1b848 |
16-Oct-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
Fix track estimation for presentation complete
Audio tracks were not using the right latency estimation for
signalling the completion of their presetation. This caused
the synchronization mechanism between playback and record to be
off, and a synchronized recording would contain some of the audio
that was meant to be over once recording would start.
Use the playback thread's latency reporting which takes the audio
pipe into account.
Bug 7237669
Change-Id: I23a907a53ad0b0d68d246789ec595a77a79fced5
udioFlinger.cpp
|
| 115b87ecf6182698b54055288d0f8710308b324d |
16-Oct-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
Fix track estimation for presentation complete
Audio tracks were not using the right latency estimation for
signalling the completion of their presetation. This caused
the synchronization mechanism between playback and record to be
off, and a synchronized recording would contain some of the audio
that was meant to be over once recording would start.
Use the playback thread's latency reporting which takes the audio
pipe into account.
Bug 7237669
Change-Id: I23a907a53ad0b0d68d246789ec595a77a79fced5
udioFlinger.cpp
|
| b9c24111ecc135d47d563eaa3609730b6b4ee5e0 |
11-Oct-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
am 0f95fadb: am 7e2c0a15: Merge "Support querying active record sources" into jb-mr1-dev
* commit '0f95fadb8a3ee402f2c51a932a06f1f685ab2d35':
Support querying active record sources
|
| aa1890ff61b06c0f19ae5cf60720e351f62ba600 |
11-Oct-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
am 0f95fadb: am 7e2c0a15: Merge "Support querying active record sources" into jb-mr1-dev
* commit '0f95fadb8a3ee402f2c51a932a06f1f685ab2d35':
Support querying active record sources
|
| d7086030fcf731e4bcef6c033cc6418cd04e6b91 |
10-Oct-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
Support querying active record sources
Add support for querying whether there is currently a recording
underway from the specified audio source.
Bug 7314859
Change-Id: I986b231a10ffd368b08ec2f9c7f348d28eaeb892
udioPolicyService.cpp
udioPolicyService.h
|
| e3f641fd9348bf7f4c07f6cc707347fccea1439c |
10-Oct-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
Support querying active record sources
Add support for querying whether there is currently a recording
underway from the specified audio source.
Bug 7314859
Change-Id: I986b231a10ffd368b08ec2f9c7f348d28eaeb892
udioPolicyService.cpp
udioPolicyService.h
|
| 1e908f4d342f7c7f0535d254837fc2e95a535650 |
09-Oct-2012 |
Eric Laurent <elaurent@google.com> |
am 80dfc5dc: am 350d3b68: Merge "audio policy: deprecate ro.camera.sound.forced" into jb-mr1-dev
* commit '80dfc5dc7ec6d20beb0360e8efcc5331f4d25d06':
audio policy: deprecate ro.camera.sound.forced
|
| 915b08fe3e65c8391a60404c52c1f87a4499eaab |
09-Oct-2012 |
Eric Laurent <elaurent@google.com> |
am 80dfc5dc: am 350d3b68: Merge "audio policy: deprecate ro.camera.sound.forced" into jb-mr1-dev
* commit '80dfc5dc7ec6d20beb0360e8efcc5331f4d25d06':
audio policy: deprecate ro.camera.sound.forced
|
| bac4a4a9073a440bc6df7ca2306604819aa1c342 |
09-Oct-2012 |
Eric Laurent <elaurent@google.com> |
audio policy: deprecate ro.camera.sound.forced
The volume and routing policy of AUDIO_STREAM_ENFORCED_AUDIBLE is
now controlled by AudioService.
Do not read ro.camera.sound.forced is not needed anymore.
Bug 7032634.
Change-Id: Ic0a6396fc4b6efb91cdb4dffe0c8eb035d0440bd
udioPolicyService.cpp
|
| 116bc97164e0fa836744f31e54779b5a7a8c9db4 |
09-Oct-2012 |
Eric Laurent <elaurent@google.com> |
audio policy: deprecate ro.camera.sound.forced
The volume and routing policy of AUDIO_STREAM_ENFORCED_AUDIBLE is
now controlled by AudioService.
Do not read ro.camera.sound.forced is not needed anymore.
Bug 7032634.
Change-Id: Ic0a6396fc4b6efb91cdb4dffe0c8eb035d0440bd
udioPolicyService.cpp
|
| 9c54cca8444bd00bba5173815d3029d45f244960 |
08-Oct-2012 |
ty.lee <ty.lee@lge.com> |
am 6f45866e: am c8823995: audioflinger/resampler: add build source for libaudio-resampler
* commit '6f45866e8a942b2a8a21c75708a717b0c8a8f120':
audioflinger/resampler: add build source for libaudio-resampler
|
| a3bdbdca6772a43cb63b70bbede8dea8359bc6cc |
08-Oct-2012 |
ty.lee <ty.lee@lge.com> |
am 6f45866e: am c8823995: audioflinger/resampler: add build source for libaudio-resampler
* commit '6f45866e8a942b2a8a21c75708a717b0c8a8f120':
audioflinger/resampler: add build source for libaudio-resampler
|
| c8823995d16b909fae30ff4f94217e875d3e8c8a |
01-Oct-2012 |
ty.lee <ty.lee@lge.com> |
audioflinger/resampler: add build source for libaudio-resampler
Bug: 7229644
Change-Id: I93bde36be1c3ec84174a4c98423e28f8b3d8782f
Signed-off-by: ty.lee <ty.lee@lge.com>
Signed-off-by: Iliyan Malchev <malchev@google.com>
ndroid.mk
|
| 2b5232dd49d006a08c35ba865a6801acdb15483c |
01-Oct-2012 |
ty.lee <ty.lee@lge.com> |
audioflinger/resampler: add build source for libaudio-resampler
Bug: 7229644
Change-Id: I93bde36be1c3ec84174a4c98423e28f8b3d8782f
Signed-off-by: ty.lee <ty.lee@lge.com>
Signed-off-by: Iliyan Malchev <malchev@google.com>
ndroid.mk
|
| ea967355de0b985c8111abd57a23f566fe355260 |
04-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
resolved conflicts for merge of f12b1c81 to master
Change-Id: I914a11706bd01a198e2137e0c12af0fc92ffa02b
|
| d192e5f15c395daa87f4f198154379ee7c93d528 |
04-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
resolved conflicts for merge of f12b1c81 to master
Change-Id: I914a11706bd01a198e2137e0c12af0fc92ffa02b
|
| ac6020508acedd316391dee42329040bf45f8d90 |
01-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Integrate improved coefficient sinc resampler: VHQ
Summary:
Very high quality is enabled only for 44.1 -> 48 or 48 -> 44.1,
and uses low quality for all other use cases.
Track estimated CPU load and throttles the quality based on load;
as currently configured it should allow up to 2 instances of very high quality.
Medium quality and high quality are currently disabled unless explicitly requested.
Details:
Only load .so the first time it is needed.
Cleanup code style: formatting, indentation, whitespace.
Restore medium quality resampler, but it is not used (see next line).
Fix memory leak for sinc resampler.
Check sample rate in resampler constructor.
Add logs for debugging.
Rename DEFAULT to DEFAULT_QUALITY for consistency with other quality levels.
Renumber VERY_HIGH_QUALITY from 255 to 4.
Use enum src_quality consistently.
Improve parsing of property af.resampler.quality.
Fix reentrancy bug - allow an instance of high quality and an instance
of very high quality to both be active concurrently.
Bug: 7229644
Change-Id: I0ce6b913b05038889f50462a38830b61a602a9f7
ndroid.mk
udioMixer.cpp
udioResampler.cpp
udioResampler.h
udioResamplerCubic.h
udioResamplerSinc.cpp
udioResamplerSinc.h
|
| a6d41334d25ffde12484eb28301352560a063ef6 |
01-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
Integrate improved coefficient sinc resampler: VHQ
Summary:
Very high quality is enabled only for 44.1 -> 48 or 48 -> 44.1,
and uses low quality for all other use cases.
Track estimated CPU load and throttles the quality based on load;
as currently configured it should allow up to 2 instances of very high quality.
Medium quality and high quality are currently disabled unless explicitly requested.
Details:
Only load .so the first time it is needed.
Cleanup code style: formatting, indentation, whitespace.
Restore medium quality resampler, but it is not used (see next line).
Fix memory leak for sinc resampler.
Check sample rate in resampler constructor.
Add logs for debugging.
Rename DEFAULT to DEFAULT_QUALITY for consistency with other quality levels.
Renumber VERY_HIGH_QUALITY from 255 to 4.
Use enum src_quality consistently.
Improve parsing of property af.resampler.quality.
Fix reentrancy bug - allow an instance of high quality and an instance
of very high quality to both be active concurrently.
Bug: 7229644
Change-Id: I0ce6b913b05038889f50462a38830b61a602a9f7
ndroid.mk
udioMixer.cpp
udioResampler.cpp
udioResampler.h
udioResamplerCubic.h
udioResamplerSinc.cpp
udioResamplerSinc.h
|
| c704ce6c0101f4933587fb2799618fb55e80f69a |
01-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
am 14bab416: am 087dd8e7: Disable audio watchdog
* commit '14bab41620ed8e667c79242e5ad0f4bd84264758':
Disable audio watchdog
|
| f1aa6eb20be602a43ea478957fc8e6d5098cfde4 |
01-Oct-2012 |
Glenn Kasten <gkasten@google.com> |
am 14bab416: am 087dd8e7: Disable audio watchdog
* commit '14bab41620ed8e667c79242e5ad0f4bd84264758':
Disable audio watchdog
|
| 0033ac1b0c2258077ad60792b5c60807b359fd0e |
01-Oct-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
am 7fff6091: am 7165491b: Merge "bug 7253033 clean up before closing an output" into jb-mr1-dev
* commit '7fff6091174451170912099954725c8109f4c559':
bug 7253033 clean up before closing an output
|
| a98a900f58d297d71621fb80afcb7d291bbb50db |
01-Oct-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
am 7fff6091: am 7165491b: Merge "bug 7253033 clean up before closing an output" into jb-mr1-dev
* commit '7fff6091174451170912099954725c8109f4c559':
bug 7253033 clean up before closing an output
|
| 087dd8e7232e4c009e9121ab7e8c37985522c9ad |
27-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
Disable audio watchdog
It's not critical, and is wasting power
Bug: 7241714
Change-Id: I6ad4375f0000c92529688723dbe0ff0caa809c5d
ndroid.mk
udioFlinger.cpp
|
| 02e5dbb025bf164f0b273f01a114ddc497798823 |
27-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
Disable audio watchdog
It's not critical, and is wasting power
Bug: 7241714
Change-Id: I6ad4375f0000c92529688723dbe0ff0caa809c5d
ndroid.mk
udioFlinger.cpp
|
| 2fa114de80a5f6afd2055054a163c6d7e2ef71bc |
28-Sep-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
bug 7253033 clean up before closing an output
An output can only be closed if there is no lock contention that
prevents ThreadBase::exit() from being blocked. If an output
device is waiting for an operation to complete (here a write
in the remote_submix module, because the pipe is full), signal
the module that it's entering the "exiting" state.
Change-Id: I8248add60da543e90c25a4c809866cdb26255651
udioFlinger.cpp
udioFlinger.h
|
| 2bfc6b42b3733c12485dd51ed95191956abc3e4e |
28-Sep-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
bug 7253033 clean up before closing an output
An output can only be closed if there is no lock contention that
prevents ThreadBase::exit() from being blocked. If an output
device is waiting for an operation to complete (here a write
in the remote_submix module, because the pipe is full), signal
the module that it's entering the "exiting" state.
Change-Id: I8248add60da543e90c25a4c809866cdb26255651
udioFlinger.cpp
udioFlinger.h
|
| 395f2c8dd0d4b87c5b31cfa838885ae8169adff8 |
27-Sep-2012 |
Iliyan Malchev <malchev@google.com> |
am 33afc99b: am 6d6a4d3f: Merge changes Ia4414241,I34f5d36a into jb-mr1-dev
* commit '33afc99b78977ab94ba3fe86f485c6e2a33563aa':
audioflinger: use resample coefficients from audio-resampler library.
audioflinger/resampler: Add VERY_HIGH_QUALITY audio-resampler
|
| f40c239e645b4e1afca3df9a022715b4056e7111 |
27-Sep-2012 |
Iliyan Malchev <malchev@google.com> |
am 33afc99b: am 6d6a4d3f: Merge changes Ia4414241,I34f5d36a into jb-mr1-dev
* commit '33afc99b78977ab94ba3fe86f485c6e2a33563aa':
audioflinger: use resample coefficients from audio-resampler library.
audioflinger/resampler: Add VERY_HIGH_QUALITY audio-resampler
|
| 41dfd129a181a5b0b62628733cf96e0ce11d64d7 |
17-Jan-2012 |
SathishKumar Mani <smani@codeaurora.org> |
audioflinger: use resample coefficients from audio-resampler library.
-Add a separate quality VERY_HIGH_QUALITY in resampler
-Use resample coefficients audio-resampler library for
quality VERY_HIGH_QUALITY.
-This improves the quality of resampled output.
Bug: 7024293
Change-Id: Ia44142413bed5f5963d7eab7846eec877a2415e4
Signed-off-by: Iliyan Malchev <malchev@google.com>
ndroid.mk
udioResampler.cpp
udioResampler.h
udioResamplerSinc.cpp
udioResamplerSinc.h
|
| 76b111685010e1fea7c0a865c038aee35507fde4 |
17-Jan-2012 |
SathishKumar Mani <smani@codeaurora.org> |
audioflinger: use resample coefficients from audio-resampler library.
-Add a separate quality VERY_HIGH_QUALITY in resampler
-Use resample coefficients audio-resampler library for
quality VERY_HIGH_QUALITY.
-This improves the quality of resampled output.
Bug: 7024293
Change-Id: Ia44142413bed5f5963d7eab7846eec877a2415e4
Signed-off-by: Iliyan Malchev <malchev@google.com>
ndroid.mk
udioResampler.cpp
udioResampler.h
udioResamplerSinc.cpp
udioResamplerSinc.h
|
| f818ccfbbfc574d179c4639af6b41c83c368a699 |
24-Sep-2012 |
SathishKumar Mani <smani@codeaurora.org> |
audioflinger/resampler: Add VERY_HIGH_QUALITY audio-resampler
Change-Id: I34f5d36ae60010ec64222d6660d10a84da3bf566
Bug: 7024293
Signed-off-by: Iliyan Malchev <malchev@google.com>
udio-resampler/Android.mk
udio-resampler/AudioResamplerCoefficients.cpp
udio-resampler/dnsampler_filter_coefficients_x128_10112011.h
udio-resampler/resampler_filter_coefficients_10042011.h
|
| 12bff5dd87e6b863482e6bfd9f4561078ebcf3f0 |
24-Sep-2012 |
SathishKumar Mani <smani@codeaurora.org> |
audioflinger/resampler: Add VERY_HIGH_QUALITY audio-resampler
Change-Id: I34f5d36ae60010ec64222d6660d10a84da3bf566
Bug: 7024293
Signed-off-by: Iliyan Malchev <malchev@google.com>
udio-resampler/Android.mk
udio-resampler/AudioResamplerCoefficients.cpp
udio-resampler/dnsampler_filter_coefficients_x128_10112011.h
udio-resampler/resampler_filter_coefficients_10042011.h
|
| 7ab148fd703107f8f4d64483d1f03986bed3f38d |
26-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
am 070936d0: am 842c5d95: Revert "Don\'t wait for presentation complete if terminated"
* commit '070936d0530b535ccc9b404872f8333368bbfb02':
Revert "Don't wait for presentation complete if terminated"
|
| fc417dc27adee8421eae1c6f506993e9a0cf2c4e |
26-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
am 070936d0: am 842c5d95: Revert "Don\'t wait for presentation complete if terminated"
* commit '070936d0530b535ccc9b404872f8333368bbfb02':
Revert "Don't wait for presentation complete if terminated"
|
| 52013e7ad02bd22b19a64bdccaa953c7b4b4d448 |
26-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
Revert "Don't wait for presentation complete if terminated"
This reverts commit 44cda3a4e7ca3de0db9cb49145def3803b03ebb4
Change-Id: I7fd29b77690dab057ac966a42fb198b2772f092c
udioFlinger.cpp
|
| 842c5d9553f3f8e97d04ed1bd0d37e4851240654 |
26-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
Revert "Don't wait for presentation complete if terminated"
This reverts commit 44cda3a4e7ca3de0db9cb49145def3803b03ebb4
Change-Id: I7fd29b77690dab057ac966a42fb198b2772f092c
udioFlinger.cpp
|
| 2de3067ebf718007c4bacd3df4fba6f070c1538b |
26-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
am 512ba315: am cc0f1cfb: Implement android.media.AudioManager.getProperty()
* commit '512ba3150d45803ebadbefe45977a5eeed323540':
Implement android.media.AudioManager.getProperty()
|
| a4b3b7df111099ae6ce4996c6ff8ea586614e0b4 |
26-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
am 512ba315: am cc0f1cfb: Implement android.media.AudioManager.getProperty()
* commit '512ba3150d45803ebadbefe45977a5eeed323540':
Implement android.media.AudioManager.getProperty()
|
| 4c6db4ce1e2760682d1c9e66448dbb39bb5d14d8 |
24-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
Implement android.media.AudioManager.getProperty()
Bug: 6635041
Change-Id: I3386a4a6c226bc4eceaf65556119e4fb15f73224
udioFlinger.cpp
udioFlinger.h
|
| cc0f1cfb69ce8b8985fc2c0984847a06a13ad22d |
24-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
Implement android.media.AudioManager.getProperty()
Bug: 6635041
Change-Id: I3386a4a6c226bc4eceaf65556119e4fb15f73224
udioFlinger.cpp
udioFlinger.h
|
| aeda50a708e41a000c509ee52cc6876ec48c191c |
25-Sep-2012 |
Eric Laurent <elaurent@google.com> |
am 4680802d: am b6ba2fd0: audioflinger: improve record start and thread exit
* commit '4680802d540379fda8a831167d8e7f2067327d90':
audioflinger: improve record start and thread exit
|
| 19dedfe04206b9d8044cb86bb2ed9dc30a26b9b5 |
25-Sep-2012 |
Eric Laurent <elaurent@google.com> |
am 4680802d: am b6ba2fd0: audioflinger: improve record start and thread exit
* commit '4680802d540379fda8a831167d8e7f2067327d90':
audioflinger: improve record start and thread exit
|
| 67d8ac3a8b49b114366684cd6310493423581056 |
25-Sep-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: improve record start and thread exit
Use broadcast() instead of signal() on the
thread wake up condition when starting record or requesting thread
exit to make sure that if another thread is waiting for the same
condition (e.g binder thread calling setParameters()) the mixer
thread will be woken up.
Bug 7184317.
Change-Id: I3154a4509ca7af6ffae5236e522b0fab8e75ed06
udioFlinger.cpp
|
| b6ba2fd0d61a4844c153e17843fbe2c841c4bf57 |
25-Sep-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: improve record start and thread exit
Use broadcast() instead of signal() on the
thread wake up condition when starting record or requesting thread
exit to make sure that if another thread is waiting for the same
condition (e.g binder thread calling setParameters()) the mixer
thread will be woken up.
Bug 7184317.
Change-Id: I3154a4509ca7af6ffae5236e522b0fab8e75ed06
udioFlinger.cpp
|
| ca301b7ec0a2a005a924d1cc7a4e988843722055 |
24-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
am 6aee72a4: am 8d452f1c: Merge "Check clock_gettime(CLOCK_MONOTONIC) for failure" into jb-mr1-dev
* commit '6aee72a479afb15498a4dc31327c5d5ff7ee4294':
Check clock_gettime(CLOCK_MONOTONIC) for failure
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| eb2ef90381ea42617f6a04b9356f5b5bf1ce4f5c |
24-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
am 6aee72a4: am 8d452f1c: Merge "Check clock_gettime(CLOCK_MONOTONIC) for failure" into jb-mr1-dev
* commit '6aee72a479afb15498a4dc31327c5d5ff7ee4294':
Check clock_gettime(CLOCK_MONOTONIC) for failure
|
| 7ca821d51972bb83fa900cf2206ae98f29b33ca9 |
24-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Check clock_gettime(CLOCK_MONOTONIC) for failure" into jb-mr1-dev
|
| 8d452f1c61aa70c4a86c4114532f2c0983120ef8 |
24-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Check clock_gettime(CLOCK_MONOTONIC) for failure" into jb-mr1-dev
|
| 99ae06be60d7ccd753a0808191583225e5533f3a |
24-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
Check clock_gettime(CLOCK_MONOTONIC) for failure
Bug: 7100774
Change-Id: I15a84a19bb6d6ef1d9dac4beaa03587638196404
astMixer.cpp
|
| 80b3273cea8660fe8a5868d024d2788a1e083ffc |
24-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
Check clock_gettime(CLOCK_MONOTONIC) for failure
Bug: 7100774
Change-Id: I15a84a19bb6d6ef1d9dac4beaa03587638196404
astMixer.cpp
|
| b777b2757b299c5fdb26d729ce405ab5b1c698ec |
24-Sep-2012 |
Eric Laurent <elaurent@google.com> |
am 842feefd: am c0c3b298: Merge "fix end of track presentation on suspended output" into jb-mr1-dev
* commit '842feefd25f3ac03de3cd9bcc5bc7627abec6ebc':
fix end of track presentation on suspended output
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| d4c1f2b55688fafabd468cd5b316dea636d7b4b8 |
24-Sep-2012 |
Eric Laurent <elaurent@google.com> |
am 842feefd: am c0c3b298: Merge "fix end of track presentation on suspended output" into jb-mr1-dev
* commit '842feefd25f3ac03de3cd9bcc5bc7627abec6ebc':
fix end of track presentation on suspended output
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| 6090a17389fc86ec6d7b16b358c540fa6627923f |
24-Sep-2012 |
Eric Laurent <elaurent@google.com> |
fix end of track presentation on suspended output
The code detecting the end of an audio track presentation before
removing it from the active track list is based on the
count of audio frames sent to audio HAL. When an output stream
is suspended (e.g. A2DP when SCO is active), this count does not
change and a track in stopped state will never be removed from
active track list causing the mixer thread to never release
the wake lock.
The fix consists in incrementing the audio HAL frame count even
if the output is suspended.
Also fix a problem in getRenderPosition() when the output is suspended.
Bug 7167534.
Change-Id: I3be836cbbea29b65dc087199cac6a1cd84c0a41d
udioFlinger.cpp
|
| 1afc26db11b71c43f63a0f72a45a803f1a7910dd |
24-Sep-2012 |
Eric Laurent <elaurent@google.com> |
fix end of track presentation on suspended output
The code detecting the end of an audio track presentation before
removing it from the active track list is based on the
count of audio frames sent to audio HAL. When an output stream
is suspended (e.g. A2DP when SCO is active), this count does not
change and a track in stopped state will never be removed from
active track list causing the mixer thread to never release
the wake lock.
The fix consists in incrementing the audio HAL frame count even
if the output is suspended.
Also fix a problem in getRenderPosition() when the output is suspended.
Bug 7167534.
Change-Id: I3be836cbbea29b65dc087199cac6a1cd84c0a41d
udioFlinger.cpp
|
| 1cbb9253100d428e52330cc71f09db08cfdcf6ae |
14-Sep-2012 |
Eric Laurent <elaurent@google.com> |
am 01dcf4de: am 896adcd3: audioflinger: send priority request from a thread
* commit '01dcf4de81ed0f9b6942dea2787463bf9c945a37':
audioflinger: send priority request from a thread
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| 6adeb50d4b320c235c4ad1e334493eb174eaeeda |
14-Sep-2012 |
Eric Laurent <elaurent@google.com> |
am 01dcf4de: am 896adcd3: audioflinger: send priority request from a thread
* commit '01dcf4de81ed0f9b6942dea2787463bf9c945a37':
audioflinger: send priority request from a thread
|
| 550f0bc9cbe8ea44b380eafa2ee674c62d10b819 |
14-Sep-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
am 868bb58e: am 4362f530: Handle audio HAL returning 0 bytes in record thread
* commit '868bb58ed5873edb1d138f82d4751823ef433ebb':
Handle audio HAL returning 0 bytes in record thread
|
| cb48dcf506e8ed9fbc5fae449ff014e495dff67d |
14-Sep-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
am 868bb58e: am 4362f530: Handle audio HAL returning 0 bytes in record thread
* commit '868bb58ed5873edb1d138f82d4751823ef433ebb':
Handle audio HAL returning 0 bytes in record thread
|
| ef6be0bb71c08db7e9e9274d70edffb95f0da25f |
13-Sep-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: send priority request from a thread
When creating a fast AudioTrack, a request is sent to SchedulingPolicyService
to elevate the requesting thread priority. This generates a binder
call into system_server process and to a JAVA service via JNI.
If the thread from which the track was created is in the system_server
process and does not have the "can call java" attribute, a crash occurs because
the binder optimization reuses the same thread to process the returning binder
call and no JNI env is present.
The fix consists in sending the priority change request from the AudioFlinger
mixer thread, not from the binder thread.
This also reverts the workaround in commit 73431968
Bug 7126707.
Change-Id: I3347adf71ffbb56ed8436506d4357eab693078a3
udioFlinger.cpp
udioFlinger.h
|
| 896adcd3ae6a1c7010e526327eff54e16179987b |
13-Sep-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: send priority request from a thread
When creating a fast AudioTrack, a request is sent to SchedulingPolicyService
to elevate the requesting thread priority. This generates a binder
call into system_server process and to a JAVA service via JNI.
If the thread from which the track was created is in the system_server
process and does not have the "can call java" attribute, a crash occurs because
the binder optimization reuses the same thread to process the returning binder
call and no JNI env is present.
The fix consists in sending the priority change request from the AudioFlinger
mixer thread, not from the binder thread.
This also reverts the workaround in commit 73431968
Bug 7126707.
Change-Id: I3347adf71ffbb56ed8436506d4357eab693078a3
udioFlinger.cpp
udioFlinger.h
|
| 3fc73b477cba93ff74e46d93ed40ebe1cfced138 |
13-Sep-2012 |
Eric Laurent <elaurent@google.com> |
am 6777e539: am f4bca03d: Merge "Don\'t deadlock on AudioRecord start when reads return 0" into jb-mr1-dev
* commit '6777e539f0479ed8ca13bfeed63edf9402b00655':
Don't deadlock on AudioRecord start when reads return 0
|
| 2fb740c2335a3321deb46b5120b77e61bf702016 |
13-Sep-2012 |
Eric Laurent <elaurent@google.com> |
am 6777e539: am f4bca03d: Merge "Don\'t deadlock on AudioRecord start when reads return 0" into jb-mr1-dev
* commit '6777e539f0479ed8ca13bfeed63edf9402b00655':
Don't deadlock on AudioRecord start when reads return 0
|
| 89b4470bc926c4b5de59deaff2b6a4993ab6cab7 |
13-Sep-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
Handle audio HAL returning 0 bytes in record thread
Enter standby when HAL returns an error, but also consider 0 bytes
returned as NOT_ENOUGH_DATA.
Change-Id: Ica83142310e9c176f936e0440571a6034cbc575f
udioFlinger.cpp
|
| 4362f5300162ed55f56cf167eccf7f2e5b89d435 |
13-Sep-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
Handle audio HAL returning 0 bytes in record thread
Enter standby when HAL returns an error, but also consider 0 bytes
returned as NOT_ENOUGH_DATA.
Change-Id: Ica83142310e9c176f936e0440571a6034cbc575f
udioFlinger.cpp
|
| 063c1eac60d7cee5d6f1ee61a10f0b6027782357 |
13-Sep-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
Fix build in AudioFlinger/FastMixer
Commit was missing some changes to modifications to send session ID
from AudioFlinger to AudioMixer.
Change-Id: I7daeaf4eff243b5287bef63be86a87acf408374a
udioFlinger.cpp
udioFlinger.h
astMixer.cpp
|
| 43ed5337f3eaa909ad2e87df0ccd061c3e6d0096 |
13-Sep-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
Fix build in AudioFlinger/FastMixer
Commit was missing some changes to modifications to send session ID
from AudioFlinger to AudioMixer.
Change-Id: I7daeaf4eff243b5287bef63be86a87acf408374a
udioFlinger.cpp
udioFlinger.h
astMixer.cpp
|
| 607364600e4f6ef8bebec8958e16e7fb0bcb7d25 |
13-Sep-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
Don't deadlock on AudioRecord start when reads return 0
When calling start() on an AudioRecord with a HAL that
returns 0 on a read() operation, the start blocking
condition was never unblocked.
Add a boolean to track the first read operation so the returned
number of bytes (mBytesRead) is only evaluated after that
first read.
Change-Id: I8c735a00d48cd6a0da467ccdf75d3616b38f6afa
udioFlinger.cpp
|
| 52762410dbc9189cd92a4094a1dbd4cfe8e71cb6 |
13-Sep-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
Don't deadlock on AudioRecord start when reads return 0
When calling start() on an AudioRecord with a HAL that
returns 0 on a read() operation, the start blocking
condition was never unblocked.
Add a boolean to track the first read operation so the returned
number of bytes (mBytesRead) is only evaluated after that
first read.
Change-Id: I8c735a00d48cd6a0da467ccdf75d3616b38f6afa
udioFlinger.cpp
|
| a59d27196cc411331efbcce5dde817c7273a64c3 |
13-Sep-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
Communicate audio session ID to downmixer
The audio downmixer effect might need the audio session Id, pass it
from the track creation in AudioFlinger to the downmix effect
creation in AudioMixer.
Change-Id: I836873eebd6711f1048fce81cd2eb29b94f0ad0c
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
udioMixer.h
|
| d06e132358d44f02c22527b4f463df1f58da9216 |
13-Sep-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
Communicate audio session ID to downmixer
The audio downmixer effect might need the audio session Id, pass it
from the track creation in AudioFlinger to the downmix effect
creation in AudioMixer.
Change-Id: I836873eebd6711f1048fce81cd2eb29b94f0ad0c
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
udioMixer.h
|
| cff7137411e737dfb783013c23841ccb335e5f5b |
11-Sep-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
Communicate audio session ID to downmixer
The audio downmixer effect might need the audio session Id, pass it
from the track creation in AudioFlinger to the downmix effect
creation in AudioMixer.
Change-Id: I5e29540542ae89cf4a0cdb537b3e67f04442a20a
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
udioMixer.h
astMixer.cpp
|
| fe3156ec6fd9fa57dde913fd8567530d095a6550 |
11-Sep-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
Communicate audio session ID to downmixer
The audio downmixer effect might need the audio session Id, pass it
from the track creation in AudioFlinger to the downmix effect
creation in AudioMixer.
Change-Id: I5e29540542ae89cf4a0cdb537b3e67f04442a20a
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
udioMixer.h
astMixer.cpp
|
| 9b827b72202038a321a9b038165971bed042311c |
11-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
am f53e33fb: am 6ce3d19c: Merge "Add outlier statistics for fast mixer cycle times" into jb-mr1-dev
* commit 'f53e33fbb03f7de07d79f5f99f73d7b2d0bbed03':
Add outlier statistics for fast mixer cycle times
|
| 240d637c7da00dd89648b96581154b514e2770e5 |
11-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
am f53e33fb: am 6ce3d19c: Merge "Add outlier statistics for fast mixer cycle times" into jb-mr1-dev
* commit 'f53e33fbb03f7de07d79f5f99f73d7b2d0bbed03':
Add outlier statistics for fast mixer cycle times
|
| 644f98718e604a58545918d99b1438e12b3a973f |
11-Sep-2012 |
Eric Laurent <elaurent@google.com> |
am 1fbb83b6: am 85396642: Merge "audioflinger: changes for new audio devices enums" into jb-mr1-dev
* commit '1fbb83b665720f2629d7a3759a0a7b61b4f04571':
audioflinger: changes for new audio devices enums
|
| 75ed25a60f65e2239c1e9e480297559825fffe8f |
11-Sep-2012 |
Eric Laurent <elaurent@google.com> |
am 1fbb83b6: am 85396642: Merge "audioflinger: changes for new audio devices enums" into jb-mr1-dev
* commit '1fbb83b665720f2629d7a3759a0a7b61b4f04571':
audioflinger: changes for new audio devices enums
|
| b89909df83e55af1726ebc97d1a5ecb4905a91ef |
09-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Add outlier statistics for fast mixer cycle times" into jb-mr1-dev
|
| 6ce3d19cd35a49add6893298f70b87917fc60d52 |
09-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Add outlier statistics for fast mixer cycle times" into jb-mr1-dev
|
| ba8a616c39259e98094440e9656a49a1cc96c976 |
07-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
Add outlier statistics for fast mixer cycle times
Change-Id: I31c964caeb8b5d9ae0a426224f030cdcb01114a0
astMixer.cpp
|
| 1ab212cf5cfa5b88c801840e11e3191cd10f48e4 |
07-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
Add outlier statistics for fast mixer cycle times
Change-Id: I31c964caeb8b5d9ae0a426224f030cdcb01114a0
astMixer.cpp
|
| 8895925e38d2521607fd2010f3f3b14ecd15962c |
28-Aug-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: changes for new audio devices enums
The ThreadBase class now has a separate member for input
and output devices (mInDevice, mOutDevice).
Only query get_supported_devices() from audio HAL if the function
is exposed and if the audio policy manager did not specify the
audio module to open.
Also fixed bug in AEC preprocessing that would reset
to default output device when an input device was given.
Change-Id: I19d4d06aeb920b068e3ef31e6e6be6345ce5d67a
udioFlinger.cpp
udioFlinger.h
|
| f1c04f952916cf70407051c9f824ab84fb2b6e09 |
28-Aug-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: changes for new audio devices enums
The ThreadBase class now has a separate member for input
and output devices (mInDevice, mOutDevice).
Only query get_supported_devices() from audio HAL if the function
is exposed and if the audio policy manager did not specify the
audio module to open.
Also fixed bug in AEC preprocessing that would reset
to default output device when an input device was given.
Change-Id: I19d4d06aeb920b068e3ef31e6e6be6345ce5d67a
udioFlinger.cpp
udioFlinger.h
|
| 6a9fa853b13e854793a9ae85adf6bb2c78c81dd5 |
07-Sep-2012 |
Eric Laurent <elaurent@google.com> |
am e1659521: am d06ab147: Merge "AudioFlinger: send audio source to audio effects" into jb-mr1-dev
* commit 'e16595211299534fe1b1986f66d55d250ffc1fea':
AudioFlinger: send audio source to audio effects
|
| ad42039eaefa06fff6cd957bcae196f087364c09 |
07-Sep-2012 |
Eric Laurent <elaurent@google.com> |
am e1659521: am d06ab147: Merge "AudioFlinger: send audio source to audio effects" into jb-mr1-dev
* commit 'e16595211299534fe1b1986f66d55d250ffc1fea':
AudioFlinger: send audio source to audio effects
|
| 2e9b0f24e7089fb23356871b2fdce771c2dae852 |
01-Sep-2012 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: send audio source to audio effects
Added support for EFFECT_CMD_SET_AUDIO_SOURCE audio effect
command to inform preprocessings of current audio source
selection for capture.
Change-Id: Ib2418a9aa8114e8457fe828ecd43b230ed86cdd6
udioFlinger.cpp
udioFlinger.h
|
| 57b2dd1e78af53115985f18d31ec5421c9da947e |
01-Sep-2012 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: send audio source to audio effects
Added support for EFFECT_CMD_SET_AUDIO_SOURCE audio effect
command to inform preprocessings of current audio source
selection for capture.
Change-Id: Ib2418a9aa8114e8457fe828ecd43b230ed86cdd6
udioFlinger.cpp
udioFlinger.h
|
| 8cd6a4a8199a84270231ca1f9212070b35e72815 |
31-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
am 08d8228d: am b3428f7f: Merge "Update audio comments" into jb-mr1-dev
* commit '08d8228d508adf26e1e6bdf44401ee3b7fcca22f':
Update audio comments
|
| 0f7d1771689280dd32eec4a301d2781fa9172baa |
31-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
am 08d8228d: am b3428f7f: Merge "Update audio comments" into jb-mr1-dev
* commit '08d8228d508adf26e1e6bdf44401ee3b7fcca22f':
Update audio comments
|
| 2662ac9220c9e07cd343e78bf3a0c21ca3e4ffd1 |
30-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Update audio comments
Change-Id: Ie7504d0ddb252f7e4d4f99ed0b44cfc7b1049816
udioFlinger.cpp
udioFlinger.h
|
| c3ae93f21280859086ae371428ffd32f39e76d50 |
30-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Update audio comments
Change-Id: Ie7504d0ddb252f7e4d4f99ed0b44cfc7b1049816
udioFlinger.cpp
udioFlinger.h
|
| fdca11917d73c21971054aa8cb9fc9ae28fa3164 |
31-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
am b5720b1f: am c9936c72: Merge "Remove dead code" into jb-mr1-dev
* commit 'b5720b1ff9262419cd331e56c75a60b291bc7694':
Remove dead code
|
| 49a3f8a8b4ec226690e7b5c91a7e516970ca6d0b |
31-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
am b5720b1f: am c9936c72: Merge "Remove dead code" into jb-mr1-dev
* commit 'b5720b1ff9262419cd331e56c75a60b291bc7694':
Remove dead code
|
| 7f7d27eb0ee5dc7b0f52dd2c89791d14bfe82f29 |
03-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
Remove dead code
RecordThread::isValidSyncEvent() returns false, so most of
RecordThread::setSyncEvent() is never executed.
Change-Id: I0cf848beb46a367a45126d2df3073c5afa2ca59c
udioFlinger.cpp
|
| 7aa25591769685ae0e8349b3ca3534c724484375 |
03-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
Remove dead code
RecordThread::isValidSyncEvent() returns false, so most of
RecordThread::setSyncEvent() is never executed.
Change-Id: I0cf848beb46a367a45126d2df3073c5afa2ca59c
udioFlinger.cpp
|
| b0dc8b57c7cc39a057170a0d1e5842bd866b1775 |
31-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
am c9beb01c: am b9cba9b7: Merge "Move libnbaio out of AudioFlinger" into jb-mr1-dev
* commit 'c9beb01c800e6326051aee52a00659288c25c6eb':
Move libnbaio out of AudioFlinger
|
| e0c87269b94e5b24de8e9577ba32070b24e83582 |
31-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
am c9beb01c: am b9cba9b7: Merge "Move libnbaio out of AudioFlinger" into jb-mr1-dev
* commit 'c9beb01c800e6326051aee52a00659288c25c6eb':
Move libnbaio out of AudioFlinger
|
| fc7992bd8220824f1404c0c54ac516d9e28b58c2 |
29-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
Move libnbaio out of AudioFlinger
libnbaio is now a separate shared library from AudioFlinger, rather
than a static library used only by AudioFlinger.
AudioBufferProvider interface is now also independent of AudioFlinger,
moved to include/media/
Change-Id: I9bb62ffbc38d42a38b0af76e66da5e9ab1e0e21b
ndroid.mk
udioBufferProvider.h
udioBufferProviderSource.cpp
udioBufferProviderSource.h
udioFlinger.cpp
udioFlinger.h
udioMixer.h
udioResampler.h
udioStreamInSource.cpp
udioStreamInSource.h
udioStreamOutSink.cpp
udioStreamOutSink.h
xtendedAudioBufferProvider.h
astMixerState.h
ibsndfileSink.cpp
ibsndfileSink.h
ibsndfileSource.cpp
ibsndfileSource.h
onoPipe.cpp
onoPipe.h
onoPipeReader.cpp
onoPipeReader.h
BAIO.cpp
BAIO.h
ipe.cpp
ipe.h
ipeReader.cpp
ipeReader.h
ourceAudioBufferProvider.cpp
ourceAudioBufferProvider.h
oundup.c
oundup.h
|
| 2dd4bdd715f586d4d30cf90cc6fc2bbfbce60fe0 |
29-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
Move libnbaio out of AudioFlinger
libnbaio is now a separate shared library from AudioFlinger, rather
than a static library used only by AudioFlinger.
AudioBufferProvider interface is now also independent of AudioFlinger,
moved to include/media/
Change-Id: I9bb62ffbc38d42a38b0af76e66da5e9ab1e0e21b
ndroid.mk
udioBufferProvider.h
udioBufferProviderSource.cpp
udioBufferProviderSource.h
udioFlinger.cpp
udioFlinger.h
udioMixer.h
udioResampler.h
udioStreamInSource.cpp
udioStreamInSource.h
udioStreamOutSink.cpp
udioStreamOutSink.h
xtendedAudioBufferProvider.h
astMixerState.h
ibsndfileSink.cpp
ibsndfileSink.h
ibsndfileSource.cpp
ibsndfileSource.h
onoPipe.cpp
onoPipe.h
onoPipeReader.cpp
onoPipeReader.h
BAIO.cpp
BAIO.h
ipe.cpp
ipe.h
ipeReader.cpp
ipeReader.h
ourceAudioBufferProvider.cpp
ourceAudioBufferProvider.h
oundup.c
oundup.h
|
| 4aa73099649056a39bd37e090c1fedd07c1ccbd5 |
22-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
am 916ecd0a: am 0c676cd0: Make tee sink compile OK again
* commit '916ecd0acb4c12568d029c027a7626fc422bd84c':
Make tee sink compile OK again
|
| 0c676cd04ff15cc7eacf6144b4aead6deaccf60c |
22-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
Make tee sink compile OK again
Change-Id: Ie865ceeabaeb698a62c11eac9f96e303c44c28ae
ipeReader.cpp
ipeReader.h
|
| b45786becf800ae7c36939c7bc92f54fe8253147 |
13-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
am a6a266d0: am 106e8a42: const methods
* commit 'a6a266d0e148f085ac1f83c4a36d3cf940c59e24':
const methods
|
| 2ef6e196198b3c2961e7fbf4637a70d6b38085ca |
13-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
am 0319e9e3: am 0dbb3560: Simplify AudioFlinger::PlaybackThread::isValidSyncEvent()
* commit '0319e9e31cb7b0033a3026b0305b78e2d48c6ec8':
Simplify AudioFlinger::PlaybackThread::isValidSyncEvent()
|
| 106e8a42038f9e90d5ff97f8ab6f1a42258bde9e |
02-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
const methods
Change-Id: I92e32ee16274c032c9d0ce910676be2a7fa52471
udioFlinger.cpp
udioFlinger.h
|
| bfb3fec3341889226c66d72ab926c7de9553a125 |
13-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
am b2f2d164: am d23eedca: Discard setSyncEvent() return value
* commit 'b2f2d1644b9d1a823fc3b78a59893652b080b35d':
Discard setSyncEvent() return value
|
| 0dbb356050d0db9e0043dd43045c1864a933332b |
03-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
Simplify AudioFlinger::PlaybackThread::isValidSyncEvent()
Change-Id: I3e4af69b929d4ca04afaac26c7e41c89fce25b9c
udioFlinger.cpp
|
| 2917080826e0d7cec47e8ec8891e9ee1c2535d6a |
13-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
am f0235625: am a9599049: Merge "Replace hard-coded "2" by a constant and comment" into jb-mr1-dev
* commit 'f02356254262ec4a56e5ec1b1abefe1849ba9f9d':
Replace hard-coded "2" by a constant and comment
|
| d23eedca9b5a1812891c05d89850ab7ee707040d |
02-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
Discard setSyncEvent() return value
setSyncEvent() returns a status_t which is sometimes ignored.
Emphasize this is intentional by casting to void.
Change-Id: Ic614988347cba36bd2504d7ad321594a355b0d9d
udioFlinger.cpp
|
| a959904968f1d300fe86334bf5099ae2dc39ee28 |
13-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Replace hard-coded "2" by a constant and comment" into jb-mr1-dev
|
| 037ac53aac0f51f2ccfaaa91822365c1c73c4365 |
10-Aug-2012 |
Jean-Baptiste Queru <jbq@google.com> |
am 0efd0358: am fcfc61da: am ff4c64d6: am c64a1787: Merge "Fix initialization of audio mixer track resources for MIPS."
* commit '0efd035854ad2c8d20100a740e054c4bf7b3e9a4':
Fix initialization of audio mixer track resources for MIPS.
|
| fcfc61da927e98d18cdcb53da8fa0a8a32ee1422 |
10-Aug-2012 |
Jean-Baptiste Queru <jbq@google.com> |
am ff4c64d6: am c64a1787: Merge "Fix initialization of audio mixer track resources for MIPS."
* commit 'ff4c64d6b1b58494a1b15f1a664e5c421d095d15':
Fix initialization of audio mixer track resources for MIPS.
|
| 3ed292031dc50c56110cdadb1e3778117e3be76a |
08-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
Replace hard-coded "2" by a constant and comment
Bug: 6679403
Change-Id: I6c2701f9afedc26540dfad0b4e23348bbc4cb01a
udioFlinger.cpp
|
| e75640c4b06a5715622eeffd8d204f811284fd8c |
09-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
am 638f936d: am 0d27c65d: Tune the overrun correction
* commit '638f936d4dd0d4afdc952988ac6b6a9dcbb577e7':
Tune the overrun correction
|
| 3c507a754788872bdc19aa651180447c5456f191 |
09-Aug-2012 |
John Grossman <johngro@google.com> |
am acc5c275: am f6a35760: Merge "AudioFlinger: fix timed audio" into jb-mr1-dev
* commit 'acc5c275f6affe789184436b7d40fb1656a4596f':
AudioFlinger: fix timed audio
|
| 0d27c65ddb5c968baa6db0c26e80f5c451bc52bc |
07-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
Tune the overrun correction
The new overrun correction is: if the previous write() cycle time is <
50% of expected, then sleep enough extra to make the total cycle time =
95% of expected. This should help compensate for HAL implementations
that have uneven cycle times due to sample rate conversion.
Bug: 6881638
Change-Id: I5ab58afdfceee7ac523177f021fbf62d743d571a
astMixer.cpp
|
| 2c3b2da3049627264b7c6b449a1622f002210f03 |
03-Aug-2012 |
John Grossman <johngro@google.com> |
AudioFlinger: fix timed audio
(cherry picked from commit e20ac92c564a2f4e8123885807abdf0a78de0dd7)
> AudioFlinger: fix timed audio
>
> Addresses Bug 6900517.
>
> Finish up support for timed audio in the new FastMixer world. Pay special
> attention to remaining lock-less and voluntary yield free on the FastMixer
> thread. This fixes audio playback for Q on JB.
>
> Change-Id: Iaf815e58a1b1d0a0190051794bec8dc5c9231785
> Signed-off-by: John Grossman <johngro@google.com>
Change-Id: I9bd687acc345a05867af48e71116690fdb0ce1b5
Signed-off-by: John Grossman <johngro@google.com>
udioBufferProviderSource.cpp
udioBufferProviderSource.h
udioFlinger.cpp
udioStreamOutSink.cpp
udioStreamOutSink.h
astMixer.cpp
onoPipe.cpp
onoPipe.h
onoPipeReader.cpp
onoPipeReader.h
BAIO.cpp
BAIO.h
ourceAudioBufferProvider.cpp
|
| ee578c0330319f04a48bccbdb26b53fea0388d04 |
24-Jul-2012 |
John Grossman <johngro@google.com> |
AudioFlinger: Better handling for master volume/mute
(cherry picked from commit 93d906837e0e89aa1d9c913ab2b531b809f9bb9e)
> AudioFlinger: Better handling for master volume/mute
>
> Changes to address bug 6842827.
>
> When a HAL is loaded, cache whether or not the HAL supports
> set_master_volume/mute in the AudioHwDevice structure. Store an
> AudioHwDevice in AudioStream(In|Out) structures instead of just an
> audio_he_device_t. This give threads (PlaybackThreads in
> particular) access to the cached capabilities.
>
> When setting master volume/mute, change the system to always set the
> setting on all HAL which support it and also to set the setting on all
> PlaybackThreads. Change PlaybackThreads to apply the setting at the
> in SW mix stage of the pipeline if its assigned HAL does not support
> the setting, or to ignore the setting of the assigned HAL does support
> it.
>
> Change-Id: Ia14137a30b4c3ee6f2d7ddcc8cba87bf5eec87f4
> Signed-off-by: John Grossman <johngro@google.com>
Change-Id: Icb6bc13764e100a2003eb1dee2231132ab287d98
Signed-off-by: John Grossman <johngro@google.com>
udioFlinger.cpp
udioFlinger.h
|
| d8f178d613821c3f61a5c5e391eb275339e526a9 |
20-Jul-2012 |
John Grossman <johngro@google.com> |
Change audio flinger to user HAL master mute if available
(cherry picked from commit 91de9b56282d126ffb36344266af5fee3cefcfdd)
> Change audio flinger to user HAL master mute if available
>
> Hand merge from ics-aah
>
> > Change audio flinger to user HAL master mute if available: DO NOT MERGE
> >
> > Replicate the pattern used for HAL master volume support to make use
> > of master mute support if the HAL supports it. This is part of the
> > change needed to address bug 6828363. Because of the divergences
> > between ICS and master, this change will need to be merged by hand.
> >
> > Signed-off-by: John Grossman <johngro@google.com>
> > Change-Id: I6d83be524021d273d093bcb117b8f2fe57c23685
>
> Change-Id: I32280582905c969aaec2bb166ec5c61df82d737a
> Signed-off-by: John Grossman <johngro@google.com>
Change-Id: I5cd709187221d307fe25c5117ccaadca5f6b197b
Signed-off-by: John Grossman <johngro@google.com>
udioFlinger.cpp
udioFlinger.h
|
| 3c0a0e8541846427db0587c2fffb90f60ee680b0 |
02-Aug-2012 |
Paul Lind <plind@mips.com> |
Fix initialization of audio mixer track resources for MIPS.
The value 1 << 32 (maxNumTracks defaults to 32) is surprisingly not defined
in C, and differs on MIPS than the other arch.
Therefore the track resources were not initialized properly, resulting
in failure to play any audio.
The fix allows the mConfiguredNames bitmask to be correctly set to all
1's in the 32-track case, for all arch.
Change-Id: Ied3e1305952e9567602e2cd76c5ef3acb0809ee0
Signed-off-by: Paul Lind <plind@mips.com>
udioMixer.cpp
|
| bdce245042dab808cb9062d9161256f01b03a79c |
01-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
am b2676a29: am 44cda3a4: Don\'t wait for presentation complete if terminated
* commit 'b2676a2920c6514ae9405cc56dfa2db49f096016':
Don't wait for presentation complete if terminated
|
| 44cda3a4e7ca3de0db9cb49145def3803b03ebb4 |
01-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
Don't wait for presentation complete if terminated
Change-Id: Ia04cf6c620693457dca87b4ffea5dd0fe71efdce
udioFlinger.cpp
|
| d4b5f7213911636a26a43127eb6ea139f403f558 |
01-Aug-2012 |
Glenn Kasten <gkasten@google.com> |
am 8d058d89: am 5665fde6: Merge "Start adding support for multiple record tracks" into jb-mr1-dev
* commit '8d058d89fd4d46bbacbb8f2dc9bd52e94cd69e56':
Start adding support for multiple record tracks
|
| 510a3d6b8018a77683dac466127ffd0af34bef6e |
16-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Start adding support for multiple record tracks
Replace single mTrack by vector mTracks.
Destroy record tracks similarly to playback tracks.
Dump all record tracks, in addition to the active record track.
Change-Id: I503f10b51928b6b92698fe1c51a9ddd3215df1f4
udioFlinger.cpp
udioFlinger.h
|
| 05144f1d68c5240a1fedd42ac83814f3ecf70dfd |
30-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Share local time frequency for all tracks & mixers"
|
| 93d906837e0e89aa1d9c913ab2b531b809f9bb9e |
24-Jul-2012 |
John Grossman <johngro@google.com> |
AudioFlinger: Better handling for master volume/mute
Changes to address bug 6842827.
When a HAL is loaded, cache whether or not the HAL supports
set_master_volume/mute in the AudioHwDevice structure. Store an
AudioHwDevice in AudioStream(In|Out) structures instead of just an
audio_he_device_t. This give threads (PlaybackThreads in
particular) access to the cached capabilities.
When setting master volume/mute, change the system to always set the
setting on all HAL which support it and also to set the setting on all
PlaybackThreads. Change PlaybackThreads to apply the setting at the
in SW mix stage of the pipeline if its assigned HAL does not support
the setting, or to ignore the setting of the assigned HAL does support
it.
Change-Id: Ia14137a30b4c3ee6f2d7ddcc8cba87bf5eec87f4
Signed-off-by: John Grossman <johngro@google.com>
udioFlinger.cpp
udioFlinger.h
|
| 91de9b56282d126ffb36344266af5fee3cefcfdd |
20-Jul-2012 |
John Grossman <johngro@google.com> |
Change audio flinger to user HAL master mute if available
Hand merge from ics-aah
> Change audio flinger to user HAL master mute if available: DO NOT MERGE
>
> Replicate the pattern used for HAL master volume support to make use
> of master mute support if the HAL supports it. This is part of the
> change needed to address bug 6828363. Because of the divergences
> between ICS and master, this change will need to be merged by hand.
>
> Signed-off-by: John Grossman <johngro@google.com>
> Change-Id: I6d83be524021d273d093bcb117b8f2fe57c23685
Change-Id: I32280582905c969aaec2bb166ec5c61df82d737a
Signed-off-by: John Grossman <johngro@google.com>
udioFlinger.cpp
udioFlinger.h
|
| 0ec23ce0d1ff79566c402bc30df3074f6e25a22b |
10-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Clean up start() parameters
Document where int is used instead of AudioSystem::sync_event_t
(probably because of a header file dependency).
TrackBase::start() and RecordTrack::start() don't need default parameters.
Change-Id: I82f4a4d078be900f3aa4bd926697e32f5ed68ec8
udioFlinger.cpp
udioFlinger.h
|
| e4e2a37dbe2a4d923232305549101f779a2e3638 |
23-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Extract methods to enter standby and standby mode
Also move initial standby from to threadLoop to avoid a race condition.
Change-Id: I65afca83c36fb41b983b3b1d3dab35d4029560e3
udioFlinger.cpp
udioFlinger.h
|
| 52008f821a5202502a82a8ba2c024e69bd336350 |
18-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Share local time frequency for all tracks & mixers
Change-Id: I5e6f0e371f728a5225c2fa6f778c109449d33602
udioMixer.cpp
udioMixer.h
|
| 0a7af18d0308295405491f86603e3d119450aba0 |
10-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Use valueAt instead of editValueAt when possible
Change-Id: I885b169f4b176a6b5c2ca9a534214b4ffff1700e
udioFlinger.cpp
udioFlinger.h
|
| 58e5aa34f01d663654d8bafad65db1dda42161ff |
20-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
effect_descriptor_t const correctness
Change-Id: Iad008f20d35a18acf500f773900164552fd0c19e
udioFlinger.h
udioPolicyService.cpp
udioPolicyService.h
|
| 1d491ff06f4b9c90ff24fe953b90d0843eaf1c04 |
16-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Fix races in AudioRecord stop()
Change-Id: Id0ac1915f57fef4a938c7f90989c1162a8b6c51c
udioFlinger.cpp
udioFlinger.h
|
| 69d799679c8c0308e42057e7b5ad63a7ae806480 |
19-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Use upmix/downmix utility routines
Change-Id: I9ae2ec938fb695ec576ea008a42205325af7bbf1
udioFlinger.cpp
|
| e65c89113232d070dd9153c54ca19301bca7a162 |
21-Jul-2012 |
Eric Laurent <elaurent@google.com> |
fix audio effect not destroyed when needed.
commit a5f44eba contained an error which made that audio effect
modules where not destroyed but left in the destroyed state
rendering them unuseable after being released.
Bug 6805168.
Change-Id: Ia4e683b3c970ffd01846c482fde73d799ff219de
udioFlinger.cpp
|
| 33e6e35b03a726e35203e97550f32154c91d5f13 |
17-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Miscellaneous audio record fixes
Call AudioSystem::stopInput() if exitPending() after wait() returns.
Acquire lock before clearing mActiveThread.
Change-Id: Ia55e4c4b3accc65ad5479cbdc094fd919152af9f
udioFlinger.cpp
|
| cd2d61016527bf48bd2e9a920bb3fdbb875eb3e4 |
19-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Use constructor to initialize instead of memcpy
Also don't check for thread parameter as it's always non-NULL
Change-Id: Id23ded1370556ef3f76f81f5f0c6fa644bcba681
udioFlinger.cpp
|
| be5f05e0fdfc4e3799653702187861a2afa072ee |
19-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Internal dump methods return void not status_t
Only the IAudioFlinger::dump() needs to return a status_t.
Change-Id: Iffeb2a7db4846df850b6b2ed960276f1fd75dba0
udioFlinger.cpp
udioFlinger.h
|
| 5ad92f620fbbb6a8281f10169a23d38e3601e07a |
19-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Revert 94479fd5405642c67efd14cebe722feb9cbe6e77
Change-Id: I5ca78d5462badf541868785b2ba2e3f6d0cf492a
udioFlinger.cpp
|
| bb4350d3b9e9485ae59e084de270f86aecef8066 |
04-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_devices_t more places
Change-Id: Id0ace02ca6f480a6c46e11880acf6cdc66d83184
udioFlinger.cpp
udioFlinger.h
|
| a189a6883ee55cf62da1d7bf5bf5a8ab501938a4 |
20-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Use struct assignment instead of explicit memcpy
for POD structs effect_descriptor_t and effect_config_t
Change-Id: Ib2fc47f85fb65ed91b0abb1f87217c49b5eb571d
udioFlinger.cpp
udioPolicyService.cpp
|
| fd4e20c226eca185fc789de761beae64855bfbbb |
04-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Run audio at priorities 2 and 3 above kernel 1
Also run the watchdog at same priority as the fast mixer.
requestPriority() originally used only the caller's pid to decide which
cgroup to assign, but in the future it might look at the priority also.
So it's safer to use same priority as the fast mixer to be sure we
run in the same cgroup.
Bug: 6461925
Change-Id: Ia59c93e4b22dacbb6746bfa6ad491be7b72f2b8d
udioFlinger.cpp
|
| d96c5724818fb47917bb5e7abe37799735e1ec0e |
25-Apr-2012 |
Glenn Kasten <gkasten@google.com> |
Don't call virtual methods in destructor
The result of calling virtual methods from a destructor is undefined.
Change-Id: I0fd4a19626e5ae564a60b753315b5f6c4b8d1f2c
udioFlinger.cpp
udioFlinger.h
|
| 1ea6d23396118a9cfe912b7b8a4e6f231e318ea2 |
09-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Use atomic ops for thread suspend count
There was a theoretical but unlikely race if two binder threads
executed suspend() or restore() concurrently. Also added comments.
Change-Id: I0908acc810b83bdd66455b27ca3429de1662a2cd
udioFlinger.cpp
udioFlinger.h
|
| 1879fff068422852c1483dcf8365c2ff0e2fadfc |
12-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Add tid parameter to IAudioFlinger::openRecord
Not yet implemented
Change-Id: I35523fb15ad71727ecc9f4bb870f07e4b7397dc4
udioFlinger.cpp
udioFlinger.h
|
| bf04a5d7f287fc712e0ed91849dc85c90c1e182d |
12-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Simplify AudioRecord::getInputFramesLost()
This also fixes a benign race in reading mActive without a lock.
Change-Id: I19e953d4f275e5c266ca1ca3fece7b6c02ad1707
udioFlinger.cpp
|
| 39c54f68804c1ce5c85ec588f3c2c63447a807b4 |
09-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Remove dead code
Change-Id: If22a6c4e572b0734eba0c5a7ce29a2c61c581e5d
udioFlinger.cpp
udioFlinger.h
|
| 4fe1ec4f40b58abff6cec147aa786cb65698161a |
28-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Fix check for invalid channel count
Change-Id: Id9e3dce0e3d5971786212d3f70e17a17e32ce92b
udioFlinger.cpp
|
| 04270daf50f0c602d7c57a257a693e68246cbeb7 |
10-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Record overflow cleanup
Add comments and rename one method for clarity
Change-Id: I04a9147e46e88a072256c0211b112d52202419e2
udioFlinger.cpp
udioFlinger.h
|
| 254af180475346b6186b49c297f340c9c4817511 |
03-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_channel_mask_t more places
Use it in AudioSystem::getOutput(), AudioSystem::getInput(),
IAudioPolicyService::getOutput(), IAudioPolicyService::getInput(),
and various other places in AudioFlinger.
Not done: AudioTrack and OutputDescriptor.
Change-Id: I70e83455820bd8f05dafd30c63d636c6a47cd172
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
udioPolicyService.cpp
udioPolicyService.h
|
| 4d2132c866228eaef563e8141ebf06aa25ee9ff7 |
10-Jul-2012 |
Eric Laurent <elaurent@google.com> |
Merge "audio policy: fix delayed command insertion"
|
| 2df8f55055fe431a508148cf525df1ba40f03113 |
10-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Add comments on use of volatile for track count
and add acquire load at the read
Change-Id: Ib41a58f5b1f6af87a8bd63d3f77d2ec0e48cb479
udioFlinger.h
|
| a34f8ec169986c5a28600c0decaa4e2db70df8e4 |
21-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Remove 'volatile' from mMute and add comments
Change-Id: I386ba27b2305a397aba70331c6bf0d35ea727cf6
udioFlinger.h
|
| 9f34a36d9cdb9595c288e50ffe00da038bc8abb9 |
21-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Whitespace and indentation
Change-Id: I424052b4ff9218147a5cfc8e6dcd67fe8105d229
udioFlinger.cpp
udioFlinger.h
|
| 7d6c35bf132a46c0a8a9826491882495fc98bd8c |
02-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Move constant initializations from onFirstRef() to constructor
Change-Id: I57f55b0bd1edee105c58c3a055f95f1e4a2c9646
udioFlinger.cpp
|
| 6648821933dc06c0b09ab2c8b32135edddcd4291 |
21-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
AudioFlinger::getBuffer() always returns non-NULL
Change-Id: I543d3db507597cacbfdad5d9ea71732137fe54fb
udioFlinger.cpp
udioFlinger.h
|
| 94479fd5405642c67efd14cebe722feb9cbe6e77 |
10-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Fix build
Revert after system/core audio_devices_t is submitted
Change-Id: I5a8ee1a7b711e834501e927f41c62efa6a6600b6
udioFlinger.cpp
|
| cec4abb2a39eafa75cf6894170a200322aa7226e |
03-Jul-2012 |
Eric Laurent <elaurent@google.com> |
audio policy: fix delayed command insertion
When inserting a delayed command in the audio policy service
command thread queue, later similar commands are filtered out.
In case of volume commands and delayed unmuting, this can cause
a problem where a new volume command can cause the timing of a
delayed unmuting command to not be honored.
This change makes that a volume command gets the time stamp of
any delayed command it replaces.
Bug 6720482.
Change-Id: I216f85ac0ea46e4a046d3483e9e9b7ffe6c34a08
udioPolicyService.cpp
|
| 01542f2704f39956da09ae2840e192dab760091f |
02-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Only write to mDevice once
This fixes a bug where readers might see intermediate values.
Also add comments about how mStandby and mDevice are used.
Change-Id: Idc84e56c21381a45137a2ca5ff9c57d437201869
udioFlinger.cpp
udioFlinger.h
|
| c1dae24a08b67b98e18e4239d4f3a74d600d353c |
03-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Remove debug code HAVE_REQUEST_PRIORITY and SOAKER
Change-Id: I73a2afe72d8acb53e57e6b4e6fb5133e22b7875a
ndroid.mk
udioFlinger.cpp
udioFlinger.h
oaker.h
|
| a5f44ebaf58911805b4fb7fb479b19fd89d2e39b |
25-Jun-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix effect disconnect deadlock
Fix possible deadlock when several EffectHandles on the same
EffectModule are destroyed simultaneously:
A wp on an EffectHandle should not be promoted to a local sp
with ThreadBase mutex held as the EffectHandle destructor can be
called when the sp gets out of scope which will call
ThreadBase::disconnectEffect() and try to acquire the mutex.
Use raw pointers instead of weak pointers for the list of handles
on an EffectModule.
Bug 6679606.
Change-Id: Ice8b602fb03a7d363c44ce3dced8a53540d96270
udioFlinger.cpp
udioFlinger.h
|
| dd8104cc5367262f0e5f13df4e79f131e8d560bb |
02-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_channel_mask_t more consistently
In IAudioFlinger::createTrack() and IAudioFlinger::openRecord(),
declare input parameter to use correct type audio_channel_mask_t.
In IAudioFlinger::getInputBufferSize(), input parameter is now channel mask
instead of channel count.
Remove unused IAudioFlinger::channelCount(audio_io_handle_t).
In AudioRecord::getMinFrameCount() and AudioSystem::getInputBufferSize(),
input parameter is channel mask instead of channel count.
Change-Id: Ib2f1c29bea70f016b3cfce83942ba292190ac965
udioFlinger.cpp
udioFlinger.h
|
| f1da96d8cf60842538e00a9c950cc451f7da2c10 |
03-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Remove longStandbyExit
It was never set (the assignment was within an "if" that was never true).
Change-Id: I01cc68e9df6b190eece621b2aa9858b4361880ce
udioFlinger.cpp
udioFlinger.h
|
| 415fa7599f48494f99206b8d6e1974abb52c5923 |
03-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Fix uninitialized field EffectModule::mPinned
Also mark EffectModule::mId and EffectModule::mSessionId const, and
document the initialization of other fields in EffectModule.
Change-Id: Ic1ca008e75e9b5924743ffc35bef80057f3a0669
udioFlinger.cpp
udioFlinger.h
|
| d5903ec1332630f2992a6f0d5ca69d13a185c665 |
18-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Compare sp<> to 0 and raw pointers to NULL
Change-Id: I50ff8a010d349d1d7e3dffa04a6331814c2128b0
udioFlinger.cpp
|
| a01992a0675a06df7d0bbe7b977207dd0c33fdc7 |
02-Jul-2012 |
Eric Laurent <elaurent@google.com> |
am dbbd5b86: am 109347d4: audioflinger: fix regression in attachAuxEffect().
* commit 'dbbd5b860a3a26bea3376410f75f27530d9cd10b':
audioflinger: fix regression in attachAuxEffect().
|
| 109347d421413303eb1678dd9e2aa9d40acf89d2 |
02-Jul-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix regression in attachAuxEffect().
Commit 717e1286 introduced a regression in PlaybackThread::Track::attachAuxEffect()
when called with an effect ID of 0 to detach the auxiliary effect.
It is normal in this case that AudioFlinger::getEffectThread_l() returns 0.
Bug 6768757.
Change-Id: I7430bd1aad2f68da38f7c3e4794e7ad657bfc6be
udioFlinger.cpp
|
| dbabf8a7dfe3aa8bf0ed169220d2009d5891fef2 |
01-Jul-2012 |
Eric Laurent <elaurent@google.com> |
am 651f9e7c: am 717e1286: audioflinger: fix auxiliary effect attachment
* commit '651f9e7c972b58a49066081187161268bcf9237a':
audioflinger: fix auxiliary effect attachment
|
| 717e128691f083a9469a1d0e363ac6ecd5c65d58 |
30-Jun-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix auxiliary effect attachment
Auxiliary effects (Reverb) are global effects and as such follow
the default rule which is to attach them to the output thread that
handles music streams by default. This causes a problem when several
threads are eligible to handle music streams as tracks can be attached
to either thread based on criteria unknown when teh effect is created.
The fix consists in moving the auxiliary effect if necessary when an
AudioTrack is attached to it and this track is not on the same
output thread.
Bug 6608561.
Change-Id: Ib32c3cabc731b2046aba728be1771982999c6069
udioFlinger.cpp
udioFlinger.h
|
| 9ee836fc8cf54d26e1e0819d1467d4d39bb9295c |
25-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove acoustics from AudioSystem::getInput()"
|
| 20010053daabfa43fcfe781bbf004473b4c08538 |
22-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Remove acoustics from AudioSystem::getInput()
Change-Id: I29fb3ee5664c1f0ee0409c1bb2be087ecca637db
udioPolicyService.cpp
udioPolicyService.h
|
| 22167855ff9af7b13fda669ca27c67a037a7d585 |
20-Jun-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix setStreamOutput()
AudioFlinger::setStreamOutput() should also work for direct outputs.
Also ignore the destination output specified to match the expected
behavior which is to invalidate all tracks using the specified stream
type so that they can be re created on the correct ouput thread.
Do not send STREAM_CONFIG_CHANGED event wich is ignored by AudioSystem
anyway since the stream to output cache has been removed.
Change-Id: I13d9d47922923b630dd755717875424c16be4637
udioFlinger.cpp
udioFlinger.h
|
| 5f12136299918ea30555f3481668ec0dbb775e5f |
15-Jun-2012 |
Eric Laurent <elaurent@google.com> |
audio policy: acquire lock for volume functions
Acquire the AudioPolicyService mutex when executing
setStreamVolumeIndex() and getStreamVolumeIndex().
These functions have become more complex and iterate on the
list of opened outputs and it is safer to acquire the
AudioPolicyService mutex to avoid concurrency with other
functions.
Bug 6626532.
Change-Id: Ifdcbbbf13338fd9a59f694d43faa7ade143f94bf
udioPolicyService.cpp
|
| eb15716b59020f342df62bce5b293f0603b94861 |
13-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Change definition of warmup period
Previously, warmup was considered done as soon as any write() took
more than 0.5 nominal cycle time. In practice, this was always the
first write() that turned on power to the output path, and it didn't
accurately account for filling the full kernel buffer queue, or for
buffering in the HAL sample rate conversion.
Now warmup is considered done when a write() _after_ the first write
takes more than 0.5 nominal cycle time.
This will throttle the initial pull rate after coming out of standby.
When combined with another change to throttle the pull rate for
devices with HAL sample rate conversion, it may help reduce some of the
notification glitches. The only downside is that it will increase the
warmup time a bit.
Bug: 6508317, 6650846, 6607056
Change-Id: I39f324c5195578170a55308e9601d3a1b41db3e0
astMixer.cpp
|
| 972af221ae7253e406e0e1ea853e56a3010ed6b1 |
14-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
FastMixer compensates for SRC jitter in audio HAL
Force a sleep if audio HAL write() returns too quickly.
Bug: 6661275
Change-Id: Ie810f5a087b89fb24dab229a0e500af1dee53571
astMixer.cpp
|
| 0a14c4ce1a41bc09eb7855fa531a3af629a69139 |
13-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Make CPU frequency statistics optional
Certain CPUs with dynamic cluster swapping and hotplug
don't report CPU frequency accurately. The file descriptors
used to read the frequency become stale and report bogus data.
So make this feature a build time option for debugging only.
This will also improve performance of the fast mixer loop.
Change-Id: I602f81ec3281a37992769208be08084ed1469e8c
ndroid.mk
astMixer.cpp
astMixer.h
|
| 362ebcbf100f7fccd37551c77e67c4faa7241b63 |
24-May-2012 |
Glenn Kasten <gkasten@google.com> |
DO NOT MERGE Remove log spam for fast track denied
Bug: 6531054
Change-Id: Iedf58e810a157aae88b5900da27c81054c437058
udioFlinger.cpp
|
| e6adde403480333b613452ae0df442df9da4a5b2 |
14-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Shorten dumpsys media.audio_flinger output" into jb-dev
|
| 49dd5cf3469cd755321f8ec10013ad4fcfaf723d |
14-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Log track name on obtain/releaseBuffer warnings" into jb-dev
|
| 3d51d5cb53cc630709a0ba78d0e60501a675f2d5 |
13-Jun-2012 |
James Dong <jdong@google.com> |
Add NOTICE and MODULE_LICENSE_APACH2 to libs build under /frameworks/av/
Change-Id: I0a3af3e2abdedebd5934f3d941d01c32cfc75e26
related-to-bug: 6647465
ODULE_LICENSE_APACHE2
OTICE
|
| 868c0abe9f26852c217bc201b337e72c68f829b8 |
13-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Shorten dumpsys media.audio_flinger output
Don't include in FastMixer if it's not present.
This removes confusing clutter especially for devices
with a separate deep buffer output thread, or for the
duplicating thread and A2DP output thread.
Change-Id: I84933f5555593256f11ba0895ec915c09cf16963
astMixer.cpp
|
| 0c9d26d187017f7fb028ab52a0fbc6395142faa4 |
31-May-2012 |
Glenn Kasten <gkasten@google.com> |
Log track name on obtain/releaseBuffer warnings
This should help diagnose problems by allowing us to correlate
the logs with the dumpsys media.audio_flinger output.
Change-Id: I8c7c592b4f87d13b0f29c66ce7a2f301a0f063c9
udioFlinger.cpp
|
| c15d6657a17d7cef91f800f40d11760e2e7340af |
30-May-2012 |
Glenn Kasten <gkasten@google.com> |
Add audio watchdog thread
Change-Id: I4ed62087bd6554179abb8258d2da606050e762c0
ndroid.mk
udioFlinger.cpp
udioFlinger.h
udioWatchdog.cpp
udioWatchdog.h
|
| 28ed2f93324988767b5658eba7c1fa781a275183 |
07-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Reduce underruns in screen off, esp. with EQ
Add MonoPipe APIs to specify setpoint.
Use screen state to configure pipe setpoint.
Fix a long-standing bug where pipe sleep time was excessive,
which interacted poorly with governor and low clock frequencies.
Now it deducts the elapsed time since last write(),
which was significant when there was EQ and low clock frequency.
Bug: 6618373
Change-Id: I6f3b0072c2244aeb033ef0795ad164491a164ff5
ndroid.mk
udioFlinger.cpp
udioFlinger.h
onoPipe.cpp
onoPipe.h
|
| a4f7e0e9a0e92a063f1b3a08988cf46e2cf1fa94 |
08-Jun-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix duplicating thread standby
The code that waits for a track presentation to be complete
before disabling it caused a regression for duplicating threads.
Because of the way output tracks activity is managed, the number
of frames output by the duplicating thread would never
reach the target set for a track to be considered presented.
The track would not be removed from active list and the thread would
not go to standby and keep its wakelock held.
Bug 6606922.
Change-Id: I4b46b420ac4cbf79a86b6791ae6589d407b01c92
udioFlinger.cpp
|
| e35a55fbdff5a695ee35ff43f1466a62fb75662b |
07-Jun-2012 |
Marco Nelissen <marcone@google.com> |
Merge "Take latency and current time into account for visualization" into jb-dev
|
| f06c2ed50e1db871ae9eb2bd15a196064f8c278c |
06-Jun-2012 |
Marco Nelissen <marcone@google.com> |
Take latency and current time into account for visualization
Buffer more data, and return the data that is currently being
output from the audio output, to ensure that visualizations are
smooth and responsive even when the audio output has a large
latency and/or large buffers.
b/6522728
Change-Id: I401637f01be7600b3c594a55c869036c13b206c0
udioFlinger.cpp
udioFlinger.h
|
| 9a0d82df8fd681257bc5b29fb5e6c65fcc7cef5a |
06-Jun-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: modify fast mixer overrun management
Do not force sleep when an overrun occurs. This was for non
blocking audio HAL and is unused.
Keep statistics on overruns.
Bug 6605215.
Change-Id: I295455d4562201aa3814c88bdd6e52c9226712af
astMixer.cpp
|
| 67c0a58e05f4c19d4a6f01fe6f06267d57b49305 |
02-May-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: various fixes on direct output
Various fixes in direct output playback thread implementation:
- threadLoop_write() was broken for playback threads that do not
use a pipe sink.
- output buffer size calculation was hard coded for stereo.
- removed software volume that was implemented for PCM stereo
format only: the audio HAL has to implement volume if needed
for direct outputs.
Change-Id: If211b4489be9af395435707b8cf0388cce1347b2
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
|
| ab9071b8d1b375418eb797c9a790da71de644344 |
04-Jun-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix active sleep time calculation.
When an audioTrack underruns, the audioflinger mixer thread
sleeps for a certain time to give the app a chance to recover.
This time is based on the reported audio HAL latency.
Some audio HALs implementing deep buffering have a variable
latency and this creates a problem if the sleep time is cached
when the output stream is opened and not updated afterwards.
This change derives the active sleep time from the mix buffer
framecount instead of the latency. This is more conservative
but works for variable latency.
Bug 6588525.
Change-Id: Ia892fc290fe06f836565c3ae15f7a2ce026c88c6
udioFlinger.cpp
|
| 399930859a75d806ce0ef124ac22025ae4ef0549 |
31-May-2012 |
Glenn Kasten <gkasten@google.com> |
State queue dump
Bug: 6591648
Change-Id: Iac75e5ea64e86640b3d890c46a636641b9733c6d
ndroid.mk
udioFlinger.cpp
udioFlinger.h
tateQueue.cpp
tateQueue.h
|
| 1295bb4dcff7b29c75cd23746816df12a871d72c |
31-May-2012 |
Glenn Kasten <gkasten@google.com> |
Fast track dumpsys
Bug: 6591648
Change-Id: I696f51c682e7233ba690d97da26012084989b412
udioFlinger.cpp
astMixer.cpp
astMixer.h
|
| bf0d21fb1310e8677caa53b90e8c3aecebc7fc13 |
31-May-2012 |
Glenn Kasten <gkasten@google.com> |
Count underruns for normal tracks also
Bug: 6591648
Change-Id: Iff9cabe392bb2ce97062603adb9c9dc7aa4170d5
udioFlinger.cpp
|
| 81787e56f8f21407eade5accc50c6da39b8af9b2 |
03-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Was calculating measured warmup time wrong" into jb-dev
|
| 7dc5b165d4f98a2059eaf1c7fec20b3b5773cd4c |
01-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix fast track leak if out of normal track names" into jb-dev
|
| 0c22272f1c97fd4b7e106030a1fd5e6b50680b61 |
01-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Pipe throttle based on requested frame count" into jb-dev
|
| 04a4ca4217e9b30342d5865024c340013d7184e8 |
01-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Was calculating measured warmup time wrong
Bug: 6592923
Change-Id: I980c6fc69d78c569147ea44ce33a7f8a41739c91
astMixer.cpp
|
| 893a05479c96f911d02beb0443da3ed6508143a7 |
30-May-2012 |
Glenn Kasten <gkasten@google.com> |
Fix fast track leak if out of normal track names
Bug: 6580402
Change-Id: I3ac7f012062c35833147f47ba822eb4bf532a824
udioFlinger.cpp
udioFlinger.h
|
| 820ba70df8ba595ae9055dfd34fdbfa32f70f14d |
31-May-2012 |
Glenn Kasten <gkasten@google.com> |
Pipe throttle based on requested frame count
Now does throttling based on the requested frame count, but
still permits overfilling pipe up to the rounded-up frame count.
Also still uses the larger frame count for non-blocking arithmetic.
Bug: 6585323
Change-Id: Ic2420a071419905524a14c22ebe1ce0e32d70a9d
onoPipe.cpp
onoPipe.h
|
| 91b14c4c144d0cc957a427cffc02ba10d0615677 |
30-May-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix effect problem during underrun
When an audio track underruns, the input buffer of the
corresponding effect chain (if any) must be cleared, otherwise
audio from previous mixer run will be fed again to the effect process
function.
Bug 6551652.
Change-Id: I5cd02196745f756c85af82d6937e9dc54369b37f
udioFlinger.cpp
udioFlinger.h
|
| f436fdcf93bd417fd3c9d2a8b19fd221d894b5e3 |
24-May-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: change session check in createTrack.
Do not refuse to create a track on an output thread if the same session
is present on another thread. It is now possible that two tracks
with the same session ID are on different threads if one can use deep
buffering and the other can't.
In this case, move effects attached to this session to the output
thread ion which the new track is created.
Bug 6530324.
Change-Id: I9019b3ee382e374c89d2319033afcfa7f886e4c4
udioFlinger.cpp
|
| e737cda649acbfa43fc1b74612a83f2fac9aa449 |
23-May-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: refine latency latency calculation.
There is an audio pipe between the normal mixer output and the fast
mixer to cope for scheduling delays and buffer size difference.
This pipe depth was not taken into account in latency calculation.
Adding the pipe contribution to the latency significantly improves A/V sync.
Bug 6520569.
Change-Id: I5584908e8aa8a02170eb38b22b4370eea800a235
udioFlinger.cpp
udioFlinger.h
onoPipe.h
|
| fbae5dae5187aca9d974cbe15ec818e9c6f56705 |
21-May-2012 |
Glenn Kasten <gkasten@google.com> |
Keep a copy of most recent audio played
Change-Id: I6b2f97881c39998a2fae9ab79d669af6c0a37e94
ndroid.mk
udioFlinger.cpp
udioFlinger.h
astMixer.cpp
astMixerState.cpp
astMixerState.h
ipeReader.cpp
|
| 6d8aabe8a3be1ac0789d00b82c3ca8b81381f5ab |
18-May-2012 |
Glenn Kasten <gkasten@google.com> |
Add throttle to have pipe fill at more stable rate
Change-Id: Ibff9ab62764e2acd04518a4a00e25f19e28aff4a
onoPipe.cpp
onoPipe.h
|
| 99c99d00beb43b939dedc9ffb07adb89f6a85ba5 |
15-May-2012 |
Glenn Kasten <gkasten@google.com> |
systrace for audio
Trace fast track buffer fill status for underruns etc.
Move the definition of macro to Android.mk.
No overhead if disabled.
Change-Id: If0e83e21b61b059ca38f543f8a6ffb58e08c79ee
ndroid.mk
udioFlinger.cpp
astMixer.cpp
|
| 552477608b137fb66af126655c6865207a82d888 |
18-May-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Display pipe underrun counters in dumpsys" into jb-dev
|
| 88cbea8a918bbaf5e06e48aadd5af5e81d58d232 |
15-May-2012 |
Glenn Kasten <gkasten@google.com> |
Display pipe underrun counters in dumpsys
The normal mixer writes it's submix to a pipe, which is read by the fast
mixer. Now dumpsys media.audio_flinger display the raw underrun counters
when fast mixer tries to pull from the pipe but doesn't get enough frames.
Change-Id: I72505f149f9e12802784da654a651d43734e1c79
udioFlinger.cpp
udioFlinger.h
|
| c059bd4246c1b3944965be921e5b334d51cd236c |
15-May-2012 |
Glenn Kasten <gkasten@google.com> |
Skip bad kHz values, and display them accurately
One device reports zero CPU kHz occasionally, so skip those values.
Preserve accuracy of the low-order 4 bits of the the CPU frequency.
Change-Id: I7d934f697412277462464fca82d112d0306a214e
astMixer.cpp
|
| 9017e5e0ebad9664bb7b6f2057e5bb29c852c64f |
15-May-2012 |
Glenn Kasten <gkasten@google.com> |
Increase normal mixer's pipe to fast mixer
Change-Id: I330925c7d07b6adb30b773bda3657e4efef9ae9b
udioFlinger.cpp
|
| 44a957f06400a338e7af20b3d16c4c4ae22a673c |
16-May-2012 |
Eric Laurent <elaurent@google.com> |
Fix static track activity ref counting
When a static AudioTrack underruns, it means that playback is over.
As apps do not necessarily stop playback explicitly, AudioFlinger
should call stopOutput() to decrease activity ref count in
audio policy manager.
Bug 6486311.
Change-Id: I1ea722c443780329ded6310c958b24726e918d16
udioFlinger.cpp
udioFlinger.h
|
| 2986460984580833161bdaabc7f17da1005a8961 |
09-May-2012 |
Eric Laurent <elaurent@google.com> |
Fix issues with synchronous record start.
- Added a timeout in case the trigger event is never fired.
- Extend AudioRecord obtainBuffer() timeout in case of
synchronous start to avoid spurious warning.
- Make sure that the event is triggered if the track is
destroyed.
- Reject event if the triggering track is in an incompatible state.
Also fix a problem when restoring a static AudioTrack after
a mediaserver crash.
Bug 6449468.
Change-Id: Ib36e11111fb88f73caa31dcb0622792737d57a4b
udioFlinger.cpp
udioFlinger.h
|
| 4adcede0dc54a85c31abaf139921aebd7a072d8e |
14-May-2012 |
Glenn Kasten <gkasten@google.com> |
Reduce video frame drop rate
The video playback engine depends on having relatively precise audio
progress updates for its A/V sync and frame drop calculations. For small
audio HAL buffer sizes, this was not a problem, but when the HAL buffer
size was > 12 ms, the normal mix buffer size became > 24 ms and this
then caused video problems. The new formula tries to keep the normal
mix buffer size within a closer tolerance of 20 ms to 24 ms.
Also use consistent term: multiplier instead of multiple.
Bug: 6479613
Change-Id: I903bad74461908e8c8f0a61e99ab5e24d5c44433
udioFlinger.cpp
|
| 09474df67278c0cd621b57c4aef1deaec4d8447f |
10-May-2012 |
Glenn Kasten <gkasten@google.com> |
Improve underrun handling for fast tracks
Maintain more accurate accounting of type of underrun.
Automatically remove track from active list after a series of "empty" underruns.
Change-Id: If042bf80e1790dcaaf195c99dc9c0ed9b55382c1
udioFlinger.cpp
udioFlinger.h
astMixer.cpp
astMixer.h
|
| d08f48c2ad2941d62b313007955c7145075d562c |
02-May-2012 |
Glenn Kasten <gkasten@google.com> |
Fix stopping process for fast tracks
Previously, the state of a fast track "wiggled" back and forth at the end.
Now it goes through these transitions:
active -> stopping_1 -> stopping_2 -> stopped
This CL is only for fast tracks, and does not change how
normal tracks work.
Change-Id: Icc414f2b48c46dda63cfa6373ca22d033dd21cd4
udioFlinger.cpp
udioFlinger.h
|
| 808e7d16504cbe5b28bb88c31afb2542ab488965 |
12-May-2012 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: fix global effects suspend logic
Audio effects on the output mix should not be suspended when effects
on the output stage (post processing) are enabled.
Change-Id: I2e1c08fa9358ea3cbaec68856738d504b1be54e4
udioFlinger.cpp
|
| e58ccce45598bcf4b4874b0e87cd1eb8d05ba9a9 |
12-May-2012 |
Glenn Kasten <gkasten@google.com> |
Fix 2 bugs in fast mixer statistics
Was losing the upper half of the bounds, and MHz was off.
Change-Id: I61c98027b35ce7c3578ea6f3acf05aa5c48a5831
astMixer.cpp
|
| d8e6fd35ec2b59ee7d873daf1f1d9d348221c7bc |
07-May-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio tag for system tracing
Disabled by default; uncomment ATRACE_TAG to enable
Change-Id: I99af894022a859ee5644bd853cfd8a48e4735ff9
udioFlinger.cpp
astMixer.cpp
|
| 42d45cfd0c3d62357a6549c62f535e4d4fe08d91 |
02-May-2012 |
Glenn Kasten <gkasten@google.com> |
Update fast mixer statistics
Compute statistics on fast mixer elapsed time and CPU load per cycle using a
simple moving average rather than by fixed blocks. This has a couple advantages:
- remove burstiness of CPU usage due to former floating-point calculations in fast mixer
- gives us flexibility in how to report (e.g. could report over just the last 1 second)
Disadvantage is increased RAM, and since the samples are not updated
atomically, it is possible to have an error in the statistics. This
should not be much of an issue due to the relatively large number of samples.
Also add CPU raw ns and adjusted MHz statistics.
Change-Id: Iaa2cd13f18250c3162aff40409b3694b769d9505
astMixer.cpp
astMixer.h
|
| e213c86d36414a8fc75e37c52999522fe09c7328 |
25-Apr-2012 |
Glenn Kasten <gkasten@google.com> |
dumpsys fCnt and flags
Previously displayed TrackBase::mFrameCount but not control block frameCount.
Now displays both.
Also display the track flags in control block.
Change-Id: Ie53781d4784633d78b6f928d69ebd494d8f110ec
udioFlinger.cpp
|
| 31dfd1db7a4d2228d9642008af6f3dd744368eb6 |
01-May-2012 |
Glenn Kasten <gkasten@google.com> |
Disable fast track log spam
except for "denied by client" and "denied by server"
Change-Id: I133ab747933729cc1f386813ee06ece055bdb294
udioFlinger.cpp
|
| 810280460da5000785662f6c5b0c7ff3ee0a4cb3 |
01-May-2012 |
Glenn Kasten <gkasten@google.com> |
Temporary fix for both normal tracks & fast tracks
If there is at least one active fast track, it forces a mixer
status of ready, which messes up the logic for normal track underruns.
Change-Id: I9de2fcaef090e2c2f99682333af3d3dd618b0d6b
udioFlinger.cpp
udioFlinger.h
|
| 288ed2103d96f3aabd7e6bea3c080ab6db164049 |
26-Apr-2012 |
Glenn Kasten <gkasten@google.com> |
Fix race condition for non-started fast tracks
This required re-implementing how fast tracks are considered active.
Now, they use the same logic as normal tracks, except underrun is ignored.
Other changes:
- add framesReady() to AudioBufferProvider interface
- rebased
- add track underrun counter state to fast mixer dump state
- move dumpsys header to Track::appendDumpHeader()
so it closer to where tracks are dumped
- display track state in dumpsys as a character code
- measure and display warmup time and cycles in dumpsys
- copy in the presentation complete code
- add ExtendedAudioBufferProvider for framesReady() which returns size_t
- simplify underrun tracking
- deferred reset track after stop()
- add comments
Change-Id: I7db8821bc565230ec76da1f9380fe3fb09735e5b
udioFlinger.cpp
udioFlinger.h
xtendedAudioBufferProvider.h
astMixer.cpp
astMixer.h
astMixerState.h
ourceAudioBufferProvider.cpp
ourceAudioBufferProvider.h
|
| 83faee053cfd4251dbb591b62039f563ffdac399 |
28-Apr-2012 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: fix stop detection for static tracks
The end of playback and end of presentation detection was broken for
static AudioTracks (tracks using shared memory buffers passed by client).
The mixer should not wait for a minimal amount of frames to be available to mix
a static track otherwise the last frames might never be consumed.
A static track should be removed from active list in case of underrun even if not
stopped().
Issue 6411521.
Change-Id: I66a2c1a77e98149e5049a223a6f04c3b8c5ad11a
udioFlinger.cpp
udioFlinger.h
|
| 300a2ee9327c05fbf9d3a5fd595b558097c7c5e8 |
25-Apr-2012 |
Glenn Kasten <gkasten@google.com> |
Fast mixer configuration
Add compile-time option for when to use fast mixer.
Double HAL frame count for fast tracks due to SRC, and make the normal
frame count multiplier an even number for compatibility. Sample rate
conversion can result in underruns if the HAL frame count is used as is,
due to jitter.
Change-Id: Ia1f8da1b8ac247d9807acfce3c318161db000905
udioFlinger.cpp
|
| da747447c1d4b5205469b4e94485b8769df57a97 |
26-Apr-2012 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: fix tracks ready for mixing logic.
Commit fec279f5 broke the logic allowing to wait for an application
to provide frames for mixing in the case of several active tracks.
This was causing audio gaps when playing music and superposing a
sound Fx (keyboard clicks...).
Issue 6185007.
Change-Id: Id0fad150d0b615646d6b1387c0de8ca944d228f6
udioFlinger.cpp
udioFlinger.h
|
| e0fa467e1150c65a7b1b1ed904c579b40f97c9df |
24-Apr-2012 |
Glenn Kasten <gkasten@google.com> |
Move frame count calculations for fast tracks
For fast tracks: move the default and minimum frame count calculations
from client to server. If accepted, the default and minimum frame count
is the fast mixer (HAL) frame count. If denied, the default and minimum
frame count is the same as it currently is for normal tracks.
For normal tracks: there is no change yet, preserve legacy behavior for
now but add a FIXME to change this later.
Bug fix: the test for buffer alignment matches channelCount was wrong.
Bug fix: check for 8-bit data in shared memory, which isn't supported.
Optimizations:
- in set(), only call AudioSystem::getOutputSamplingRate() when needed
- in createTrack_l(), only call AudioSystem::getSamplingRate() and
AudioSystem::getFrameCount() when needed
Change-Id: I79d2fe507db1a8f7bb094c71da8a129951dbb82f
udioFlinger.cpp
|
| 1dc28b794587be22c90a97070d928f94586db638 |
24-Apr-2012 |
Glenn Kasten <gkasten@google.com> |
Use scheduling policy service
Change-Id: I3c09da1dc0de5039d0c15ce7fb2bc373fa398712
ndroid.mk
udioFlinger.cpp
SchedulingPolicyService.cpp
SchedulingPolicyService.h
chedulingPolicyService.cpp
chedulingPolicyService.h
|
| 58912562617941964939a4182cda71eaeb153d4b |
03-Apr-2012 |
Glenn Kasten <gkasten@google.com> |
AudioFlinger normal mixer uses FastMixer
Change-Id: I3131bb22d2d057e9197a2ebfa6aa1cfaab9e5321
ndroid.mk
udioFlinger.cpp
udioFlinger.h
oaker.h
|
| 3acbd053c842e76e1a40fc8a0bf62de87eebf00f |
28-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Configure policy of mediaserver threads
Change-Id: Ifd825590ba36996064a458f64453a94b84722cb0
ndroid.mk
udioFlinger.cpp
udioFlinger.h
|
| c95cfbb87d0ac5e773037019a96bfc29972d4b4e |
12-Apr-2012 |
John Grossman <johngro@google.com> |
TimedAudioTrack: Optimize the queue trim operation.
Hand merge from ics-aah
> TimedAudioTrack: Optimize the queue trim operation.
>
> Don't perform the end PTS calculation for each buffer during trimming.
> Instead, only calculate the ending PTS of a buffer if there is no next
> buffer in the queue. This optimization assumes that the buffers being
> queued are in monotonic media time order (a fair assumption for now)
> and that the timestamps in the audio are contiguous (not a requirement
> for this API, but a reality of how it is being used right now).
>
> In the case where the audio is discontinuous on purpose, it is
> that this optimization will cause the system hold one extra buffer
> which it could have safely trimmed. It should not be much of an issue
> since in real life the audio is almost always contiguous, and as long
> as the media clock is running and the mixer is mixing, the buffer will
> be used up and discard as part of the normal flow anyway.
>
> Change-Id: I00061e85ee7d5651fcf80751646c7d7415894a14
> Signed-off-by: John Grossman <johngro@google.com>
Change-Id: I0054b58e1389fa005aa990cb5710caf4af7b706a
Signed-off-by: John Grossman <johngro@google.com>
udioFlinger.cpp
|
| 8d314b709fdd81bb64bdaa8d72a0b19c355cefb9 |
19-Apr-2012 |
John Grossman <johngro@google.com> |
TimedAudioTrack: Fix continuity threshold handling.
Hand merge from ics-aah
> TimedAudioTrack: Fix continuity threshold handling.
>
> Fix issues with continuity threshold handling; notably
> + If the steady-state continuity threshold is exceeded, be sure to
> clear the on-time flag. Failure to do this will result in the
> system picking a new mix point which simply satisfies the
> steady-state continuity threshold instead of the startup threshold.
> Since we are putting a discontinuity in presentation anyway, we
> really want to pick a perfect point, not just an OK point.
> + Tighten the steady-state continuity threshold. It was currently set
> to 100mSec which is enormous. 4mSec (the new setting) is much more
> appropriate. On systems with a VCXO (like tungsten) this should
> never be wrong by more than a sample. If TimedAudioTracks are ever
> to be used on VCXO-less systems, this threshold should probably be a
> a parameter configurable by applications on a track by track basis
> so they can make the tradeoff between allowed error and frequency of
> disruptive corrections.
> + Reset the on-time flag if the mixer provides no PTS during a mix
> operation. This makes for a convenient way for the HAL to reset
> timed tracks when it makes changes for delay compensation across
> multiple outputs.
>
> Change-Id: I2cb23de5a3d1f75618abc1c8ab903db883837aa8
> Signed-off-by: John Grossman <johngro@google.com>
Change-Id: Ibd28c9d290494b0b19eb01caf2d9bfdef606a9b4
Signed-off-by: John Grossman <johngro@google.com>
udioFlinger.cpp
|
| 21e8c50bd13ebe44f3088e26c9c6df0e163c469c |
12-Apr-2012 |
Glenn Kasten <gkasten@google.com> |
FastMixer update
Updates:
- Add support for mono fast tracks
- Add support for optional sample rate conversion on fast tracks
- Log sample rate and frame count
- Enable statistics
Change-Id: Ife014edf4f452da361f3eaaae19209ef6ff6958b
ndroid.mk
astMixer.cpp
astMixer.h
astMixerState.cpp
astMixerState.h
|
| 9bd23229fdec1657398abc682ccccfce1c95f8aa |
16-Apr-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
Fix multichannel downmix pause bug on video player
Pausing a video player will cause the track audio mixer to be
disabled, which causes the downmixer to be deleted. When reenabled,
the track channel mask hasn't changed but the downmixer is there
anymore.
Fixed by:
- instanciating a downmixer when the AudioMixer track
gets initialized (in getTrackName(), now taking a channel mask
as input), and deleted when in deleteTrackName().
- when the channel changes on a track, check whether it
needs a downmixer or not. Preparing a track for downmix
automatically removes the old downmixer if there was one.
Also: initialize the track downmixerBufferProvider field
when AudioMixer is instanciated, so we can safely call
delete on it in AudioMixer's destructor, in case
deleteTrackName() wasn't called before the mixer was
destroyed.
Change-Id: I589b0781cda5b3c82f85b561c52b08546cac21f8
udioFlinger.cpp
udioMixer.cpp
udioMixer.h
astMixer.cpp
|
| 0ca3cf94c0dfc173ad7886ae162c4b67067539f6 |
18-Apr-2012 |
Eric Laurent <elaurent@google.com> |
rename audio policy output flags
Change-Id: I27c46bd1d1b2b5f96b87af7d05b951fef18a1312
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
udioPolicyService.h
|
| acb86cccbd9d245439a04cef0bcefa589addaa4c |
16-Apr-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
Configure the resampler with the correct channel count when downmixing
When a track needs to be downmixed and resampled, it gets downmixed
first before being resampled. Therefore the resampler needs to
be configured with the channel count of the output of the downmixer
instead of that of the track.
Removed frame size checks that don't apply anymore now that we support
frame sizes that are not powers of 2 (e.g. 12 for 5.1 16bits), and
changed test performed for every buffer during playback into an
assert.
Change-Id: Ia220f00ee382f4f7848b661c58555bdca664e194
udioFlinger.cpp
udioMixer.cpp
|
| 97b5d0d5b5ef766eb5dd680d05a5d199662d4ae0 |
24-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Fast mixer
Change-Id: I61552f83507e08e4c706076b9fb15362869e6265
ndroid.mk
astMixer.cpp
astMixer.h
astMixerState.cpp
astMixerState.h
tateQueueInstantiations.cpp
|
| 0845682ff93c9765b74b53dc404d0954efc28c2e |
18-Apr-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Add template class StateQueue"
|
| dc998c809e084b617990b281e2ed5271830cc2e0 |
24-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Add template class StateQueue
Change-Id: Iccc5eb42bc295a22b2e429a4551f083cd7b6831a
ndroid.mk
tateQueue.cpp
tateQueue.h
|
| f7ffb8bf0a58037f0bc9662c5275005a4e539948 |
14-Apr-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: update to new audio HAL.
Updated audio flinger to new function prototypes
for open_output_stream() and open_input_stream().
Check audio hw device version when loading a module and
reject devices with a version different from current one.
Change-Id: I9d4c81a1f59a15db78b3989417c2a411c638fe88
udioFlinger.cpp
|
| 4e2293f29f2e719af1245d365747ea06d074b345 |
12-Apr-2012 |
Glenn Kasten <gkasten@google.com> |
AudioMixer new cmd: remove sample rate converter
Add a new command REMOVE to remove any sample rate converter
on a track, without having to delete the track name.
Add comments.
Remove some dead code.
Fix whitespace and comparison to NULL.
Change-Id: Id55a23ab5ee673189e99675b4e6fa5df7b617987
udioMixer.cpp
udioMixer.h
|
| a5e821439996de6005b2fa36b3bdd31f003ce23f |
16-Apr-2012 |
Eric Laurent <elaurent@google.com> |
AudioMixer: fix crash in destructor
Commit deeb1282 made that track->resampler is only set to NULL when
getTrackName() is called. It remains uninitialized for all unused tracks
causing a crash when deleted by Audiomixer destructor.
Issue 6309441.
Change-Id: I49451387a6afe81299b8368ef206524df32f74c4
udioMixer.cpp
|
| b388e531613730572067e193b6b66afb5d042233 |
14-Apr-2012 |
Eric Laurent <elaurent@google.com> |
Fix ALOG_ASSERT in AudioFlinger
Fix broken ALOG_ASSERT in updateFramesPendingAfterTrim_l() introduced by
commit 1c345196.
Change-Id: Ie1b2653069283f23ff0367f2628828e37fb0749c
udioFlinger.cpp
|
| d3030da2ac3c0ebb8b7bdf38418263caf405b863 |
12-Apr-2012 |
John Grossman <johngro@google.com> |
Fix the build
forgot to upload final fixup during merge. sry about that
Change-Id: I2ddd2c08d8efa83c0a8d1e378ae4c28686145154
udioFlinger.cpp
|
| 1c345196edc61694f29307a1826a64a0d26028dc |
27-Mar-2012 |
John Grossman <johngro@google.com> |
TimedAudio: Track of the number of pending frames.
This is a manual merge from ics-aah
> TimedAudio: Track of the number of pending frames.
>
> Keep track of the number of frames pending in the timed audio queue so
> we can implement framesReady in O(1) time instead of O(N). This
> change partially addresses bug 6020970; the bug will be completely
> addressed once this change has been up-integrated into master.
>
> Change-Id: I599eb15ea1f6d715b97b30e65214fb6fadd169df
> Signed-off-by: John Grossman <johngro@google.com>
Change-Id: I6cbbbc3afc8efd066fe94865326ede0c6b3db2bd
Signed-off-by: John Grossman <johngro@google.com>
udioFlinger.cpp
udioFlinger.h
|
| 9fbdee13d09447550dd22ae72c2dbabdce7f0a80 |
27-Mar-2012 |
John Grossman <johngro@google.com> |
TimedAudio: Fix a cause of audio popping.
This is a manual merge from ics-aah
> TimedAudio: Fix a cause of audio popping.
>
> Fix an issue with buffer lifecycle management which could cause audio
> pops on timed outputs. There were two issues at work here.
>
> 1) During trim operations for the queued timed audio data, buffers
> were being trimmed based on their starting PTS instead of when the
> chunk of audio data actually ended. This means that if you have a
> very large chunk of audio data (larger than the mixer lead time),
> then a buffer at the head of the queue could be eligible to be
> trimmed before its data had been completely mixed into the output
> stream, even though the output stream was fully buffered and in no
> danger of underflow.
> 2) The implementation of getNextBuffer and releaseBuffer for timed
> audio tracks was not keeping anything like a reference to the data
> that it handed out to the mixer. The original architecture here
> seemed to be expecting a ring buffer design, but timed audio tracks
> use a packet based design. Pieces of packets are handed out to the
> mixer which then frequently will hold onto that chunk of data
> across two mix operations, using the first part of the chunk to
> finish a mix buffer and then using the end of the chunk for the
> start of the next mix buffer. If the buffer that the mixer is
> holding a piece of got trimmed before the start of the next mix
> operation, it would return to its heap and could be filled with who
> knows what by the time it actually got mixed. On debug builds,
> they seem to get zero'ed out as they go back to the heap causing
> obvious pops in presentation.
>
> This change addresses both issues. Trim operations are now based on
> ending presentation time for a chunk of audio, not the start. Also,
> when the head of the queue is in flight to the mixer, it can no longer
> be trimmed immediately, merely flagged for trim by the mixer when the
> mixer finally does call releaseBuffer.
>
> Signed-off-by: John Grossman <johngro@google.com>
> Change-Id: Ia1ba08cb9dea35a698723ab2d9bcbf804f1682fe
Change-Id: I2c5e2f0375c410f0de075886aac56ff6317b144c
Signed-off-by: John Grossman <johngro@google.com>
udioFlinger.cpp
udioFlinger.h
|
| 3f9c84c0a5af83fceb8669390e2d71b75ec7b550 |
04-Apr-2012 |
Eric Laurent <elaurent@google.com> |
audio pre processing: test code for dual mic
Added functional test code to validate effect API for
multi mic simplementations.
Also fixed warning in AudioFlinger.
Change-Id: I07be4d2e4d17791d3626c804ba3e9f87ff26d05a
udioFlinger.cpp
|
| 7d5b26230a179cd7bcc01f6578cd80d8c15a92a5 |
05-Apr-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
AudioMixer uses downmix effect for multichannel content
In the AudioMixer structure associated with each track, add an object
that acts as the buffer provider when the track has more than two
channels of input in the mixer. This object, DownmixerBufferProvider,
gets audio from the actual buffer provider of the track, and applies
a downmix effect on it.
The downmix effect is created and configured when the track gets
created in AudioFlinger, which causes AudioMixer::getTrackName()
to be called with the new track's channel mask. It is released
when the track is disabled in the mixer.
Change-Id: I05281ed5f61bef663a8af7ca7d5ceac3517c82db
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
udioMixer.h
|
| 3f502bfc6d04d08369cf147284c5c4c2ac7c9a0e |
06-Apr-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Non-blocking audio I/O interface, WIP"
|
| a4c5a550e2a3bc237179b8684e51718e05894492 |
29-Mar-2012 |
Eric Laurent <elaurent@google.com> |
audio policy: load audio hw modules.
Audio HW modules are now loaded upon request from audio policy manager
according to the configuration in audio_policy.conf.
Removed hard coded HW module loading by AudioFlinger at init time.
Added methods to IAudioFlinger and AudioPolicyInterface
to control the loading of audio HW modules.
Added methods to open an output or input stream on a specific hw module.
Change-Id: I361b294ece1a9b56b2fb39cc64259dbb73b804f4
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
|
| 1a9ed11a472493cac7f6dfcbfac2064526a493ed |
21-Mar-2012 |
Eric Laurent <elaurent@google.com> |
audio policy: add configuration file
removed outputs to stream mapping cache in audio system: the output for a
given stream type must always be queried from audio policy manager as the cache
is not always updated fast enough by audioflinger callback.
removed AudioFlinger::PlaybackThread::setStreamValid() not used anymore if
stream to output mapping is not cached.
Change-Id: Ieca720c0b292181f81247259c8a44359bc74c66b
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
|
| 73d227557ba5192735356bacab9f77b44980793b |
19-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
AudioFlinger track flags and server's fast policy
Change-Id: I72358c8e6829d173b3e60ced8a8babc089869fac
udioFlinger.cpp
udioFlinger.h
|
| 0bf65bdde04b8e66c998ff37e2b2afafddddfa33 |
29-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
const methods and comments
Change-Id: Ifd16750174fdb15b72507787502b587562ffc99e
udioFlinger.cpp
udioFlinger.h
|
| 010662326b9c43c703725f933e95e0897f8a6bdd |
27-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Non-blocking audio I/O interface, WIP
Yet another abstraction similar to AudioTrack::Buffer and AudioBufferProvider,
but with support for streaming, non-blocking, and eventually PTS.
This is intended to be used as follows:
- primary HAL output stream implements a Sink
- primary HAL input stream implements a Source
- Pipe implements a Sink
- PipeReader implements a Source or TimedSource (not shown yet),
which supports "read at PTS"
- fast AudioTrack on server side will implement a Source using cblk
- normal AudioTrack on server side will not be changed initially
- fast AudioRecord on server side will implement a Sink using cblk
- normal AudioRecord on server side will not be changed initially
- fast mixer thread will read from Sources and write to a Sink,
or (unlikely) implement a Source and multiple Sinks
- Visualization and PCM logger will read from Source or TimedSource
- A2DP normal mixer will be connected directly to its output stream
and there will be a kind of OutputTrack for duplication that will
read from a Sink with non-blocking write fed by the fast mixer.
Patch set 3 changes:
- Add more implementations of NBAIO interfaces:
added SourceAudioBufferProvider, MonoPipe, MonoPipeReader.
- Added Format_sampleRate and Format_channelCount.
- Extract out the roundUp() method.
- Respond to most comments from previous code review.
- The new classes are untested.
Patch set 4 changes:
- Fix bugs in MonoPipe::write() and MonoPipeReader::read()
- Fix bug initializing mFrameBitShift too early
- renamed roundUp() to roundup()
- Fix Android.mk
- Add LOG_TAG an LOG_NDEBUG, use ALOG_ASSERT and utils/Log.h instead of assert
- Fix build warnings
- Move constructor and destructor bodies from .h to .cpp
- Line length 100
- Following naming conventions for #include double-include protector macros
- Include what you use
- More NBAIO logging
- MonoPipe write can be blocking
Patch set 5 changes:
- Address code review comments
- Use a static library so unused implementations don't take memory
- Comment out libsndfile dependency
- Remove debugging LOGV and LOG_NDEBUG
Patch set 6 changes (would be 6 at old location, actually 2 at new location):
- Address code review comments on patchset 5
- For MonoPipe, allow the full pipe to be used, no need to omit one slot
- Don't do atomic releasing stores unless needed
Still to do:
- I'm not happy with the Pipe class names
- Update build/ for new static library?
Change-Id: Ie6c61f05ce06b676b033be448a8ef9025a2ffcfd
ndroid.mk
udioBufferProviderSource.cpp
udioBufferProviderSource.h
udioStreamInSource.cpp
udioStreamInSource.h
udioStreamOutSink.cpp
udioStreamOutSink.h
ibsndfileSink.cpp
ibsndfileSink.h
ibsndfileSource.cpp
ibsndfileSource.h
onoPipe.cpp
onoPipe.h
onoPipeReader.cpp
onoPipeReader.h
BAIO.cpp
BAIO.h
ipe.cpp
ipe.h
ipeReader.cpp
ipeReader.h
ourceAudioBufferProvider.cpp
ourceAudioBufferProvider.h
oundup.c
oundup.h
|
| deeb1282621f3177ad667360b40eef8e4fedb298 |
25-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
AudioMixer default track parameters
Move the default initialization of track parameters from the AudioMixer
constructor to getTrackName(). This fixes a bug where the defaults were
only valid the first time a name was allocated.
Also added a switch case for parameter FORMAT.
Change-Id: I07abd3aaf7f4efe9825a761d2bc662aaee2c6db0
udioMixer.cpp
|
| a1472d9883e35edd280201c8be3191695007dfd4 |
30-Mar-2012 |
Marco Nelissen <marcone@google.com> |
Make AudioTrack/AudioRecord handle more than 2^32 frames
b/6160363
Change-Id: I471815012c6a113ec2c4dd7676e8fa288a70bc76
udioFlinger.cpp
|
| a011e35b22f95f558d81dc9c94b68b1465c4661d |
30-Mar-2012 |
Eric Laurent <elaurent@google.com> |
implemented synchronous audio capture
Added the infrastructure to support the synchronization of playback and
capture actions on specific events.
The first requirement for this feature is to synchronize the audio capture
start with the full rendering of a given audio content.
The applications can further be extended to other use cases
(synchronized playback start...) by adding new synchronization events and
new synchronous control methods on player or recorders.
Also added a method to query the audio session from a ToneGenerator.
Change-Id: I51f1167290d9cafdf2fbcdf9e4785156973af44c
udioFlinger.cpp
udioFlinger.h
|
| b83d38feeeb88a8a2a6219e1fca2480b5a14fb0d |
26-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "IAudioFlinger::createTrack and openRecord flags"
|
| d12c68ad699ce0ed822a4d4db86e8e02324c6b03 |
23-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Revert "AudioFlinger does not need libmedia any more"
This reverts commit c920dee060ac69684be33210ee44b99a5fc3e8b2
ndroid.mk
|
| 4f5da11df06802856bf526f16563df7d5755f653 |
22-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
AudioFlinger does not need libmedia any more
Change-Id: Ifd2c61882109ec36ca68072a2bf6506e08c8cf34
ndroid.mk
|
| c5c49398584f2399af905a931e556ed6e0a29cd4 |
21-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Clean up Track constructor"
|
| ccf89b54f973f11082150d02ed957f7e967fbc8b |
21-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioMixer can be configured for fewer max tracks"
|
| 63c1faa8dea7feb90255d31ef2a133d8f2818844 |
21-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Update comments"
|
| 7153494670bdac8b650cb10b8b1838651e0ca418 |
21-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Whitespace"
|
| a26ff6f22f4e86d09514c2819237bd9748455018 |
21-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "new doesn't fail on Android"
|
| 9a5f413a21b320fea0607e653bc75b1a4f0e7a2e |
21-Mar-2012 |
Eric Laurent <elaurent@google.com> |
am a48285c4: am 165ee4c5: am 14958e21: Merge "audioflinger: fix issue with camcorder and A2DP" into ics-mr1
* commit 'a48285c4f22ffc43f1771ebd1ff35dcec48db2c7':
audioflinger: fix issue with camcorder and A2DP
|
| 5c94b6c7689a335e26a86e8a0d04b56dc627738e |
21-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
AudioMixer can be configured for fewer max tracks
Change-Id: I371b17cef071d083eecf35cd3627a3adff907a33
udioMixer.cpp
udioMixer.h
|
| f99590187e2e3f1cf6f093063170edec269cac5d |
19-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Clean up Track constructor
The 'thread' parameter can never be NULL.
Use constructor initialization list when possible.
Make more members const.
Only put the relevant code under "if (mCblk != NULL)".
Add comment about track name leak.
Change-Id: Ib963390a69bed1999638cc982a759edd1d5f4712
udioFlinger.cpp
udioFlinger.h
|
| ea7939a079b3600cab955760839b021326f8cfc3 |
14-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Whitespace
Fix indentation, and add blank lines in key places for clarity
Change-Id: I57a0a8142394f83203161aa9b8aa9276abf3ed7c
udioFlinger.cpp
udioMixer.cpp
|
| 2ee367e444e7b62e02bde8a2e47603a9ad342c6e |
20-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Add libmedia_native
Change-Id: I3ac357c78fb89f108d15c6e5b9fa317de0e9fb9a
ndroid.mk
|
| 17a736c3e1d062d7fc916329eb32aef8935614af |
14-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Update comments
Change-Id: I327663a020670d0a72ff57bd0b682e2ce0528650
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
udioMixer.h
|
| a03567676e8766828ff970b87e13bc4c97b23473 |
19-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
new doesn't fail on Android
Change-Id: I5079a3bf31097dd0807b2d806d5f8d3cff2077ab
udioFlinger.cpp
|
| a075db4ff9b086ac2885df77bb6da0869293df92 |
06-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
IAudioFlinger::createTrack and openRecord flags
createTrack and openRecord don't need the "old" flags parameter,
which was either audio_policy_output_t or audio_in_acoustics_t
shifted left by 16 bits. But they do need "new" flags, which
are defined by the application use case. Initially, the only
application use case flag is timed output, but others are planned.
For output, the audio_policy_output_t flags are passed to
AudioSystem::getOutput, which returns an audio_io_handle_t, and that
handle is then passed to createTrack. So createTrack doesn't need the
old flags parameter.
For input, the audio_in_acoustics_t flags are passed to
AudioSystem::getInput, which returns an audio_io_handle_t, and that
handle is then passed to openRecord. So openRecord doesn't need the
old flags parameter.
Change-Id: I18a9870911846cca69d420c19fe6a9face2fe8c4
udioFlinger.cpp
udioFlinger.h
|
| 9d7b4c074205609271f61e1a4741ac0c524a1795 |
19-Mar-2012 |
Eric Laurent <elaurent@google.com> |
am 14958e21: Merge "audioflinger: fix issue with camcorder and A2DP" into ics-mr1
* commit '14958e21c12f922d7501d32c3bec05109eb342d5':
audioflinger: fix issue with camcorder and A2DP
|
| 89d94e79dad032fb18ddc655e6068e4231d3f0aa |
17-Mar-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix issue with camcorder and A2DP
Some audio HALs do not support well a device selection of 0 (no device)
received on an input stream.
This can happen because of a problem in the audioflinger code that handles
the forwarding of the output device selection to the record thread for use by
the pre processing modules that need it. If the output device is 0 (meaning
no op, which happens when stopping playback over A2DP) audioflinger could not
detect it was an output device selection and would forward it to the input
stream (see AudioFlinger::setParameters() and RecordThread::checkForNewParameters_l().
Issue 6179641.
Change-Id: Idae534521866538e0d12ba259a2834f402a922e2
udioFlinger.cpp
|
| b12153e36ae5e514d96c77d660b327d1a8f5000f |
16-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Add libmedia_native
Change-Id: Ib8cff8abd73723b793f08da99ad59549f219e0e7
ndroid.mk
|
| 33b383948e8f270bff30378476f00dce289004eb |
13-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Remove dependency on audio_* location
Change-Id: I4bc66115fcb9ba22b057bd72db3f561dcb18a0d8
ndroid.mk
|
| dfaf549e3e310bc22444f4404b19f4907b24c286 |
15-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioFlinger playback thread CPU measurement in Hz"
|
| 190a46f7c84e160386610c0c4cecb9767fb5503b |
06-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
AudioFlinger playback thread CPU measurement in Hz
Log statistics on CPU usage in Hz in addition to wall clock time
Use CPU statistics for all playback threads, not just MIXER
(but they are disabled by default by a compile-time debug macro).
ThreadCpuUsage library:
- Move statistics out of the library and leave that up to the caller
- Add API to determine a CPU's frequency
Change-Id: Ia1011123146e641fcf210ef26e78ae2b4d3b64ad
udioFlinger.cpp
|
| 3b229ed97c0dfc85a8cf881341e29e595e0edea7 |
14-Mar-2012 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: more info in dumpsys"
|
| 612bbb57c59397a540e96f06bdd16e437a583af5 |
14-Mar-2012 |
Eric Laurent <elaurent@google.com> |
audioflinger: more info in dumpsys
Added TID and io handle to AudioFlinger threads dump.
Change-Id: Ib1a856f3bad55c73e4c395b5e59d57435f4b9a4c
udioFlinger.cpp
|
| d3cee2f0f649c01e1153d593cbe723887b8e0ba0 |
14-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Break circular dependency on media player service
Bug: 6165157
Change-Id: I3c85bbcaf31f3cb9a009e273f7b6284015eb3bd8
udioFlinger.cpp
|
| e53b9ead781c36e96d6b6f012ddffc93a3d80f0d |
13-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Whitespace and indentation
Fix indentation to be multiple of 4.
Make it easier to search:
sp< not sp < to
"switch (...)" instead of "switch(...)" (also "if" and "while")
Remove redundant blank line at start or EOF.
Remove whitespace at end of line.
Remove extra blank lines where they don't add value.
Use git diff -b or -w to verify.
Change-Id: I966b7ba852faa5474be6907fb212f5e267c2874e
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
udioPolicyService.h
udioResampler.cpp
udioResampler.h
udioResamplerCubic.h
udioResamplerSinc.cpp
|
| fd83fbf5ead098070bae674b20e6f87f45ab5d4c |
13-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Clean up assertion checks"
|
| aa4397f07c43bd83bc3100b749401dc3d15e7622 |
13-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Fix bug where mMixerStatus was set to IDLE
Change-Id: I55df1738fb7ba17ba6caeea6a17557526eac17a7
udioFlinger.cpp
|
| 5798d4ebf236357a4b13246f40e52b90a34d09a4 |
08-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Clean up assertion checks
Use ALOG_ASSERT instead of assert.
Use compile-time asserts where appropriate.
Fix typo in an ALOGV.
Change-Id: I58f1c1ffc14319a022c88b5a88b8d0368660da8b
udioFlinger.cpp
udioMixer.cpp
|
| 73f4bc33e0d458933460250a47c64aa868c05f97 |
09-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Inline applyVolume() into threadLoop_mix()
Also the declaration of applyVolume in PlaybackThread was dead.
Change-Id: I4b1a9848d07d3d7f340baea05b17f667c78df868
udioFlinger.cpp
udioFlinger.h
|
| 470aa50c36089fbe0427557f7cf4464dd26a1c52 |
12-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove unnecessary friend declarations"
|
| 2d6ef93773465cd3e66146fac35050a472c589f7 |
12-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove virtual from methods that don't need it"
|
| 66fcab972e9218d47c58a915f391b2f48a09903a |
24-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge dup code at thread entry and param change
This CL is mostly just cleanup, but there are a couple of fixes marked
"FIX" below.
Merge the duplicate code that was at the beginning of threadLoop() and
after a parameter change. cacheParameters_l() is now called at entry to
threadLoop() and after any parameter change. It re-calculates all values
that are derived from parameters, and caches them in instance variables.
updateWaitTime_l():
- FIX activeSleepTime depends on mWaitTimeMs, which was initially set
to infinity. updateWaitTime_l() was not called at entry to
threadLoop(), so activeSleepTime was not set correctly before the
first parameter change.
- FIX reversed the order of calls after parameter change
for the same reason so that updateWaitTime_l() is called before
calculating values that are derived from wait time.
- marked it private since now it's only called from DuplicatingThread
Change-Id: If2607d2ed66c6893d910433e48208a93c41fb7e9
udioFlinger.cpp
udioFlinger.h
|
| 18868c5db2f90309c6d11e5837822135e4a0c0fa |
07-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_policy_output_flags_t consistently
This affects:
- IAudioFlinger::openOutput
- AudioTrack::AudioTrack
- AudioTrack::set
- apps that call these
Change-Id: I26fb281bac6cb87593d17697bc9cb37a835af205
udioFlinger.cpp
udioFlinger.h
|
| d69549665d412f1f6ebad48ad8cd05133ada8728 |
09-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Replace hard-coded 3 by FCC_2 to simplify searches"
|
| 688aac7675f18bdd7bff13334759e20bc4e6c390 |
09-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Remove virtual from methods that don't need it
Change-Id: I30e17e61aae25b036436c0e270313c80c43e5f06
udioFlinger.h
|
| 1998661fdb6b0b5ae103e047e3d653c5da1b99e3 |
09-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Remove unnecessary friend declarations
Add comments to the remaining friends, so we know what is left if we
decide to remove them later.
Change-Id: I1de929257dc4700960f77902cda3d303177c72cf
udioFlinger.h
|
| 083c154162c88a9f63aeaa10a4b52dd454bda9ff |
09-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Don't ask policy manager about invalid stream type"
|
| b3dfcf483a96816fe2e949dd86fed26e52ad7404 |
09-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioPolicyService InputDesc minor cleanup"
|
| a19ffb656616feec70613ba67ddfe15a504a4e76 |
09-Mar-2012 |
Eric Laurent <elaurent@google.com> |
Merge "audio policy: use audio_devices_t when appropriate"
|
| 81872a2ce65653ae980621ba2907da56a83ba9a7 |
08-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
AudioPolicyService InputDesc minor cleanup
Add a non-default constructor to set the mSessionId, and make mSessionId const.
Remove explicit clear on mEffects - it is automatically cleared by the destructor.
AudioPolicyService::setPreProcessorEnabled:
- parameter is const *
- use an alias instead of making a Vector copy
Destructor doesn't need to be virtual since there are no subclasses.
Change-Id: Ibc3c3bea8259839430b1cf5356186c7d96f1082f
udioPolicyService.cpp
udioPolicyService.h
|
| 53d76dbe7c55821e89d9da02e7a563f7fb45de87 |
08-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Replace hard-coded 3 by FCC_2 to simplify searches
Change-Id: I92881d04e8378307f849fb343071a58d181a68b4
udioFlinger.cpp
udioFlinger.h
|
| 6374252107fd6539397598195ea6defd5870fafb |
08-Mar-2012 |
Eric Laurent <elaurent@google.com> |
audio policy: use audio_devices_t when appropriate
Change-Id: I1b3a5879e81c789fb53d356af3d3a1ee2dca955f
udioPolicyService.cpp
udioPolicyService.h
|
| fec279f5a0bfaa2a42e91ab6dfa0282baeee308b |
08-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Mixer status cleanup
Use mPrevMixerStatus for DirectOutputThread also.
Remove the MIXER_CONTINUE logic and use MIXER_IDLE instead.
Rename the field mixerStatus to mMixerStatus.
Rename local variable back to mixerStatus.
Change-Id: I0a8145fc856c6c5ff8b784b6176ef3c4d8eb7408
udioFlinger.cpp
udioFlinger.h
|
| b071e9bc248865ef87a339044c0c5cbabfac175c |
08-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Cleanup DirectOutputThread::mActiveTrack
Rename activeTrack to mActiveTrack.
Release the reference earlier, at the end of threadLoop_mix().
This allows the field to be made private and to
move the declaration from PlaybackThread to DirectOutputThread.
Change-Id: I02be7a254638f7d85e92aaf0002d20ca0092a5c3
udioFlinger.cpp
udioFlinger.h
|
| 639dbee79140956c43926344c23af765f6e0c9a5 |
07-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Don't ask policy manager about invalid stream type
Change-Id: If50fbff9d34045d1398984da48da7e6428a74491
udioFlinger.cpp
|
| b279312a9038b9c5b9b05b31b1b1db86f748efd8 |
08-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "IAudioFlingerClient::ioConfigChanged param2 const"
|
| fa26a859bacb597587a8f49acfc3127351fa688c |
06-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Isolate references to outputTracks/mOutputTracks
Move all references to DuplicatingThread::outputTracks and
DuplicatingThread::mOutputTracks from the common threadLoop() into
virtual methods. This allows them to be moved from PlaybackThread to
DuplicatingThread, and to be marked private.
Also use vector assignment to copy mOutputTracks to outputTracks.
Change-Id: Ieb1cf1ad36b8a65143e61e6c92a65fb43427e5e2
udioFlinger.cpp
udioFlinger.h
|
| a13ad6e84db128eadf23b154d3346f0bb473a5f7 |
07-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Make applyVolume private to DirectOutputThread"
|
| d4513b09123deebf8023b73a82d2d46d35806cea |
07-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Make applyVolume private to DirectOutputThread
Change-Id: I7ca4a59505857cbd106b6f274c66e9580dead271
udioFlinger.cpp
udioFlinger.h
|
| a3707a280177e934a1e0a15660d9176663b7fc17 |
07-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Add comments about sequence for setting parameters"
|
| 1465f0c3df04c3166155a852a6a5c10069c1fd0a |
06-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge the calls to prepareTracks_l
Change-Id: I1dd759581333e2908d980180d44db7bf5ed6591d
udioFlinger.cpp
udioFlinger.h
|
| b81cc8c6f3eec9edb255ea99b6a6f243585b1e38 |
01-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
IAudioFlingerClient::ioConfigChanged param2 const
The 3rd parameter (param2) to AudioFlingerClient::ioConfigChanged
is used as an input. So changed it from void * to const void *.
It is then cast to const OutputDescriptor *
or const audio_stream_type_t * depending on the event.
Change-Id: Ieec0d284f139b74b3389b5ef69c7935a8e5650ee
udioFlinger.cpp
udioFlinger.h
|
| f8edf68a1e39da273eafe8c85bdbedc26636c4ec |
07-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Rename fields of AudioSessionRef"
|
| cbc52bfb3b51c81c945b1e35990324bde892829a |
01-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Add comments about sequence for setting parameters
Change-Id: Iffa59a34c3c47bdc1d3234cdcb4d8ff99c102825
udioFlinger.h
|
| 92b8360fe9c3174dc0edaaab4b746d8a3d3f831f |
07-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix indentation for re-organized code"
|
| 438b036deaf4413503567a75a2861c000d21da5b |
06-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Rename updateWaitTime since a lock is held
Change-Id: I9bb978cbd0debf5b21676467060f72eebafea3e6
udioFlinger.cpp
udioFlinger.h
|
| 952eeb27682a9b2ddfa761f24b6eb5e18fe5d814 |
06-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Fix indentation for re-organized code
Change-Id: I63471cebdbd095b7ad4e481611b785f9b02c7941
udioFlinger.cpp
|
| 012ca6b4f69fb24385025c0e84b8f816525a3032 |
06-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Rename fields of AudioSessionRef
Change-Id: I9f2a66094135c4ac6bec2d3e9db3ac5fbf988ede
udioFlinger.cpp
udioFlinger.h
|
| 000f0e39b4d0c88441297a05ab5f8da6832c1640 |
02-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
threadLoop merge
Change-Id: Id8e6330ac6be76f9c2debba94f856de87e2d98f7
udioFlinger.cpp
udioFlinger.h
|
| 3030fcefcfcfc48789612659206b957cd0489a54 |
01-Mar-2012 |
Eric Laurent <elaurent@google.com> |
renamed audio policy output flag.
Renamed AUDIO_POLICY_OUTPUT_FLAG_INDIRECT to AUDIO_POLICY_OUTPUT_FLAG_NONE
which is more appropriate.
Change-Id: Ia14d60397df0f2dcd9bea0186400a09da35bc104
udioPolicyService.h
|
| e8286332f3817a8b7cc4cfd8f6450a3913533660 |
29-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Shorten thread names"
|
| 3e07470f3b122097cacfe5b85cdb1359279a2f33 |
29-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Prepare for threadLoop merge - active tracks
Continued work on making the copies of threadLoop more similar:
- Remove alias for mActiveTracks in MixerThread and DuplicatingThread.
- Pull in declaration of activeTrack in DirectOutputThread.
- Remove redundant parameter of prepareTracks_l().
- Comment prepareTracks_l().
Change-Id: If1087c1902b454acec01ddfdd9f055f0ca7abf04
udioFlinger.cpp
udioFlinger.h
|
| bc4b08ba67a0245e092aee8f08ba30ef22d421bf |
29-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Update AudioFlinger comments"
|
| 91cda8bdf8358a8b977a44e016b60a70bc1a5ee9 |
29-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Pull in declaration of effectChains to inner block"
|
| 73ca0f5837d5448f7a5eb159a09cd0ebe82b4de9 |
29-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Pull in declaration of effectChains to inner block
Change-Id: I09eacf72124942abd604132b9f4e774b1236fcf3
udioFlinger.cpp
|
| c455fe9727d361076b7cead3efdac2d32a1a1d6d |
29-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
mSuspend comments and usage
Emphasize that playbackthread::mSuspend is a counter, not a bool
Change-Id: I7188e56814e1c54dbc65e560f3627f138257d644
udioFlinger.cpp
udioFlinger.h
|
| 688a64030834ea2f52cc9765676ddf6aa34df767 |
29-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Mark similar and different sections in threadLoop
Most of these comments will be removed after the threadLoop merge.
Note: the trivial change in assignments to mixBufferSize, and the
comments about "tracks to remove" is to make them all identical.
Change-Id: I3b1a33a7f2cd12ad557a1986bb71f6171161974a
udioFlinger.cpp
|
| d805b718b1fd2d5407ef665c8d4bb42e63dc71a9 |
25-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Update AudioFlinger comments
Add comments to enum mixer_state
Note side-effect of lockEffectChains_l
Fix a typo
Change-Id: Ibd51678bac2193201cbcbe081ff5664046fbc494
udioFlinger.h
|
| 480b46802bef1371d5caa16ad5454fce04769c57 |
28-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Shorten thread names
prctl(PR_SET_NAME) limits to 15 characters. Before we had names like
"Binder Thread #" and the counter was cut off :-( Also remove redundant
"thread" at end of name; it's always a thread.
Change-Id: I1f99c2730ba0787ed9b59c15914356cddf698e2f
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
|
| a3873833d518e032138cf70188b6f33cd7acec3d |
28-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Unlock effect chains in the middle of two if's"
|
| 3e9c3a1d34960cd258f294d31135ab6bf76179d5 |
28-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Simplify removeNotificationClient"
|
| a17c820c556fddf7ddd96b82b3e9874e340ffafd |
28-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioFlinger const methods and parameters"
|
| cfbd62616ab2b12f0fee603658f04e5827cc7f8f |
28-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix theoretical race condition in addOutputTrack"
|
| fadb2c73fce479205432652530663e1e90fd546c |
28-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioBufferProvider comments and cleanup"
|
| 843a12d146bd64642bf85a4e56c274246e3893a6 |
27-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix tracking of hardware state for dump"
|
| e628d515888baadba75442128678e747e930ed58 |
27-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Make threadLoop() logs identical"
|
| d3cee0b1f77baa4fb7a049eb757e9f5006890726 |
27-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Move declaration of mixerStatus to inner block"
|
| a3b09254d44cd8d66ec947abe547538c4cfeaa89 |
20-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Simplify removeNotificationClient
No need to check for presence of item before removing
(but we do lose the log of the previous value).
Change-Id: I2838430824de5f257f2ee15db0c22b1920c67d08
udioFlinger.cpp
|
| 02fe1bf923bbe5789202dbd5810e2c04794562e6 |
25-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
AudioFlinger const methods and parameters
Change-Id: I93ec28024005ed23aa141518092a012a4a7c44c5
udioFlinger.cpp
udioFlinger.h
|
| c0b52836d07f823732f0ff98ca5ca9d7f5730cb8 |
24-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Make threadLoop() logs identical
Change the wording of the logs in the various copies of threadLoop()
to be identical. This will make it easier to merge them soon.
Change-Id: Idfa181e437738712c784dc7f746cac79f83d2931
udioFlinger.cpp
|
| 5d4eeeaf76ebe177b43e87b2a9df5e55e39021f0 |
24-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Move declaration of mixerStatus to inner block
mixerStatus was being declared (and initialized) too early,
which also resulted in a duplicate initialization. Moved
the declaration into the block where it is actually used.
Change-Id: Ifdcfefe362a5efe3493dd616cdb44645c6f9aed5
udioFlinger.cpp
|
| 37d825e72a6c606553a745da1212590a425996d3 |
24-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Pull out duplicated copies of silent mode check
Also fix the error handling for the property_get.
This is part of preparation for the threadLoop() merge.
Change-Id: I6405190ea18146d1271575e1dfe9f279e8f36b17
udioFlinger.cpp
udioFlinger.h
|
| 04743e99e71c0da012508c7119f414027654ee94 |
24-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Unlock effect chains in the middle of two if's
As part of the upcoming threadLoop() merge, this CL makes it clearer
what are the similar and different parts before and after unlocking
effect chains.
In each threadLoop(), the old code was:
if (sleepTime == 0) {
// A
unlockEffectChains(effectChains);
// B
} else {
unlockEffectChains(effectChains);
// C
}
The new code is:
if (sleepTime == 0) {
// A
}
unlockEffectChains(effectChains);
if (sleepTime == 0) {
// B
} else {
// C
}
Also this is slightly slower by one "if", it has the advantage of making
it much more obvious about what is done before and after the unlock,
and also to see the similarities and differences among the various
copies of threadLoop().
Change-Id: I7bf4369d2dcb072573ec43b7e52c637f0097dc00
udioFlinger.cpp
|
| 5ce96d97feafc6989f6141bb2633eae3d87ddf28 |
24-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Pull CPU statistics code out of threadLoop()"
|
| b6b740629c9f11535086e744465bada03f26df11 |
24-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Fix theoretical race condition in addOutputTrack
This is not a real race, because addOutputTrack was only called in two
places, and in both places there could be no other threads referencing
the DuplicatingThread instance.
Those two places are:
- the DuplicatingThread constructor, which is of course safe
- openDuplicateOutput - this is safe because it's called immediately
after the new DuplicatingThread, and there are no sp<> either in the
constructor or here which could cause onFirstRef() to do Thread::run().
But for safety in case addOutputTrack is ever called somewhere else,
or there are sp<> created earlier, it is safer to take the thread lock.
Change-Id: I1502d014fa37ec5dbf4bf40d3e2884af311cd5e9
udioFlinger.cpp
|
| 83efdd0fc08cd5aedf50b45741a8a87be8dc4b41 |
24-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Pull CPU statistics code out of threadLoop()
This is to prepare for the threadLoop() merge
Change-Id: I118c7d5c6b011b5d5b95ec7d63fb03feb166a9cf
udioFlinger.cpp
|
| 01c4ebf6b794493898114a502ed36de13137f7e5 |
22-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
AudioBufferProvider comments and cleanup
Add comments about which methods implement the AudioBufferProvider interface.
Simplified the definition of kInvalidPts. <stdint.h> is very hard to work
with, there seems to be no way to use it reliably to get INT64_MAX without
having a separate source file, which is ugly because it means kInvalidPts
is not a compile-time constant. So I just deleted AudioBufferProvider.cpp
and used a hard-coded constant instead.
Added a default constructor for Buffer so that the fields aren't random
(especially .raw which is used to determine if the buffer is valid).
Make the pts for getNextBuffer default to kInvalidPTS so code that
doesn't need a pts doesn't have to specify a value.
Rename the parameter to AudioMixer::setBufferProvider to make it clearer.
Change-Id: I87e7290884d4ed975b019f62d1ab6ae2bc5065a5
ndroid.mk
udioBufferProvider.cpp
udioBufferProvider.h
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
|
| 8abf44d2f2bcd20a2835570efe89d89c19db426a |
02-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Fix tracking of hardware state for dump
At end of AudioFlinger::onFirstRef(), the hardware status was being left
in wrong state. It should be AUDIO_HW_IDLE but was AUDIO_HW_INIT.
mHardwareStatus was being set to AUDIO_HW_OUTPUT_OPEN too early, and so
a return would leave it in the wrong state until next hardware operation.
Take the hardware lock for dev->get_parameters, and update mHardwareStatus
before and after.
Keep hardware lock only for the duration of the dev->set_parameters.
Rename two constants in enum hardware_call_state to have the prefix
AUDIO_HW so they follow the naming conventions.
Add comments.
Change-Id: I6c7450b11f9b13adaeef9cec874333e478a58fc0
udioFlinger.cpp
udioFlinger.h
|
| 5cf034d92d901169ca6e36c90475f40715827fcd |
21-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Remove TrackBase::mFlags
The bit-field TrackBase::mFlags was supposed to have track-specific
flags in the upper 16 bits, and system flags in the lower 16 bits.
The upper 16 bits of mFlags were initialized in the TrackBase
constructor from the flags parameter of IAudioFlinger::createTrack()
and IAudioFlinger::openRecord(), and the lower 16 bits were cleared.
However, the upper 16 bits of mFlags were never acccessed again.
So really there are no track-specific flags. I left the flags
in the parameter list of createTrack() and openRecord() but made a
note that these should be removed eventually as they are dead.
This leaves only the one system flag "step server failed". I replaced
the bit-field mFlags by bool mStepServerFailed, which is simpler and
slightly faster.
Change-Id: I6650f5487be72791b4a67d73adcd10ffa04e2aa5
udioFlinger.cpp
udioFlinger.h
|
| d6fd85a157ce2054b2304e6d171fa87ae09c363d |
22-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Avoid wp<>::unsafe_get() with a few exceptions"
|
| 9eaa55756c5b245970447019250ce852f5189525 |
20-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Avoid wp<>::unsafe_get() with a few exceptions
Avoid using wp<>::unsafe_get() except in a log, and other specific cases
when it's known to be safe.
Use more specific subclass types for parameters to avoid down-casts.
When a constructor or method parameter is "this" of an object that is
currently being constructed, it's better to use a raw pointer rather
than either sp<> or wp<>.
Using the raw pointer is safe, provided either:
- it is "this" of an object being constructed (which has sp<> refcount of 0),
- or the caller already holds an sp<>
The raw pointer is simpler and faster, and it avoids the problem of the
sp<> reference count being incremented and then decremented to zero on
scope exit, which would cause the object's destructor to run while the
object is still being constructed.
Also removed some dead code per a review comment.
Change-Id: I7375f64da3aec11b928c33cb01faff186252ef5e
udioFlinger.cpp
udioFlinger.h
|
| f063b49e95c28d63a58215ebda892a5fee4204cc |
18-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Fix build warning
warning: pointer of type 'void *' used in arithmetic
warning: enumeral and non-enumeral type in conditional expression
Change-Id: I7b8d626a636145ef648e3b5d0e77068216dd012e
udioFlinger.cpp
|
| 3b81acab52b7140c1b8b20be2d67be3e221637e7 |
28-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Remove bit fields to improve performance
uint16_t enabled is (mostly) changed to bool in a separate CL
Change-Id: Ied9f8c034b2479cee9a8778cee7b8ff92ae75b7b
udioFlinger.cpp
udioMixer.cpp
udioMixer.h
|
| 1b094ee8f7fe7eca65bf3d2f983ba95eef6db93d |
17-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Simplify code"
|
| a111792f1314479c649d1d44c30c2caf70c00c2a |
26-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Simplify code
Use DefaultKeyedVector::valueFor to avoid extra test
Make local variables as local as possible
No double parentheses
No typedef for single use
No parentheses around indirect function call
No AudioFlinger:: prefix when not needed
Remove unnecessary casts
Remove block with only one line
Saves 128 bytes
Change-Id: I3a87430eeb01b81e7b81a1c38f6fdd3274ec48f3
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
udioMixer.h
udioPolicyService.cpp
|
| 29dcfcd66d884801e9907d04e81d407ee770802c |
17-Feb-2012 |
Mike Lockwood <lockwood@google.com> |
Merge "Put a bandaid on a segfault in timed audio track handling."
|
| fe5b3ba4b332d5fc9aa4f453434329b9f38768c2 |
13-Feb-2012 |
John Grossman <johngro@google.com> |
Put a bandaid on a segfault in timed audio track handling.
Add a bandaid to prevent a segfault which can occur while handling
timed audio buffers. There is a deeper problem which should
eventually be addressed, but for now this fix should prevent any
crashing.
The deeper problem is as follows.
When the AudioFlinger mixer gets data to mix from an AudioTrack, it
ends up getting a structure filled out which points into an IMemory
region owned by the AudioTrack. Unfortunately, this structure is not
holding a refcount on the IMemory which it points into. If the
IMemory refcount hits 0 and the chunk of RAM is retuned to the binder
heap it came from, there can still be a Buffer object being held by
the AudioFlinger mixer which points into the region of memory which
was retuned to the binfer heap. If AF reads from this buffer, it
could read corrupt data (if the region of memory gets handed back out
to a writer), or it could segfault (if the heap has been freed and the
pages unmapped). Similar problems could happen if AF attempts to
write to the buffer, heap corruption in one case, segfaulting in the
other.
In the past, this has not been an issue for AF, because tracks
allocate a single IMemory (which serves as a ring buffer) and the
IMemory lives for as long as the track lives. As an artifact of the
way the code came out, the mixer cannot be holding a Buffer structure
pointing into the IMemory which used to be owned by a track if the
track no longer exists. Tracks cannot come into or out of existence
during a mix operation, which is the only thing which makes this safe.
TimedTracks work differently, however. Timed tracks each allocate a
small binder heap, and then hand out IMemory instances broken out of
this heap. The heap lives as long as the track, so the worst which
could happen here is that a TimedTrack's IMemory gets returned to the
heap while there is still a buffer structure in flight pointing into
the memory region, then the region gets handed out again and
overwritten by new data causing the mixer to mix the wrong audio. The
timing to cause this to happen is very difficult to encounter, and you
to generate the timing conditions required, you need to be in a pretty
bad failure state where audio is already breaking up and skipping, so
its unlikely that anyone would notice (which is why I'm band-aiding
the segfault and letting the deeper issue slide for now).
In general, however, it might be a good idea to revisit this buffering
design. On principal, if someone is going to hold pointers into a
refcounted object, they should be holding a ref on the object at the
same time. Failure to do this will usually lead to a situation where
there are corruption or segfault issues, or to a system where the
refcounted object's lifetime must be implicitly managed very carefully
in ways which are usually non-obvious and are easy to break by new
engineers on a project.
Change-Id: Ib391075395ed0ef46a03c37aa38a82d09e88abeb
udioFlinger.cpp
|
| 9fda4b87441fe17d90d8144639c9de6d9022c3c0 |
02-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Fixed possible heap corruption in EffectDesc
"EffectDesc *effect = new EffectDesc(*effects[i]);" was relying on the
default copy constructor for EffectDesc, but the default copy constructor
does a member-by-member copy. This works OK for mUuid, but a member
copy of mName and mParams shares pointers. This could result in heap
corruption later on due to a double free. Changed to add an explicit
copy constructor that does a deep copy of both mName and mParams.
A malloc() and strdup() were being freed by delete, but the correct
matching API for these is free(). Fortunately our current memory runtime
implementation ignores the difference. Changed to use free().
EffectDesc and InputSourceDesc member fields were being torn down by
the code that does delete. Changed to do the tear-down in ~EffectDesc()
and ~InputSourceDesc().
Added constructor EffectDesc() with name and UUID parameters, rather
than having caller fill in the object after construction.
Made ~EffectDesc() and ~InputSourceDesc() non-virtual to save memory,
since they have no subclasses.
Change-Id: Ibb5cc2e6760d72e0c4cf537068ac4432c717bafd
udioPolicyService.cpp
udioPolicyService.h
|
| ef7740be67a4d7b6b033ebed59c3d4a9c74a2c18 |
09-Feb-2012 |
John Grossman <johngro@google.com> |
Fix a segfault in AudioFlinger.
Check the string returned by a HAL's implementation of get_parameters
for NULL before attempting to make use of it. That way, we won't
bring down the mediaserver because of a poorly written HAL.
Change-Id: Ic99d7b004520d7d6347842a681c0595e889b68ea
Signed-off-by: John Grossman <johngro@google.com>
udioFlinger.cpp
|
| 4ff14bae91075eb274eb1c2975982358946e7e63 |
09-Feb-2012 |
John Grossman <johngro@google.com> |
Upintegrate Audio Flinger changes from ICS_AAH
Bring in changes to audio flinger made to support timed audio tracks
and HW master volume control.
Change-Id: Ide52d48809bdbed13acf35fd59b24637e35064ae
Signed-off-by: John Grossman <johngro@google.com>
ndroid.mk
udioBufferProvider.cpp
udioBufferProvider.h
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
udioMixer.h
udioResampler.cpp
udioResampler.h
udioResamplerCubic.cpp
udioResamplerSinc.cpp
|
| 6dad4378f2a78d967defc8912ecf47f6ed117584 |
14-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix races related to volume and mute"
|
| d9b9b8d09e7471b0ffa21cfa9f944ef4ad300a71 |
14-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Update comments"
|
| 99e53b86eebb605b70dd7591b89bf61a9414ed0e |
19-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Update comments
We no longer put the filename at start of file.
Change-Id: Ic435b159a23105681e3d4a6cb1ac097bc853302e
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
udioMixer.h
|
| 8d6a2449a91f5116d7243ab039393195ebd663fe |
08-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Use size_t and ssize_t with Vector
Use size_t with size() and ssize_t with indexOfKey(). Exception:
use ssize_t for backwards loops, and indices that are overloaded as a
marker or error code.
Change-Id: Ibf2a360af4539b72b09c818dda22ea2a0de92431
udioFlinger.cpp
udioPolicyService.cpp
|
| 6dbc1359f778575d09d6da722b060a6d72c2e7c5 |
02-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
AudioRecord and AudioTrack client tid
Inform AudioFlinger of the tid of the callback thread.
Change-Id: I670df92dd06749b057238b48ed1094b13aab720b
udioFlinger.cpp
udioFlinger.h
|
| 44deb053252a3bd2f57a007ab9560f4924f62394 |
06-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Factor out and speed up permission-checking code
Use the caching permission check for dump to save IPC.
Cache getpid() to save kernel call for other permission checks.
The C runtime library getpid() can't cache due to a fork
race condition, but we know that mediaserver doesn't fork.
Don't construct String16 on the stack.
Change-Id: I6be6161dae5155d39ba6ed6228e7683e67be34ed
ndroid.mk
udioFlinger.cpp
udioPolicyService.cpp
erviceUtilities.cpp
erviceUtilities.h
|
| 2b213bc220768d2b984239511cd4554a96bc0079 |
02-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
mAudioHwDevs and related cleanup
Inline AudioFlinger::initCheck and remove unnecessary lock.
Remove redundant check of mAudioHwDevs.size().
No need to lock mHardwareLock for each device separately
during initialization.
Use size_t not int to loop through Vector, since size() returns size_t.
Add missing hardware lock for get_mic_mute() and get_input_buffer_size().
Add comments.
Change-Id: Iafae78ef78bbf65f703d99fcc27c2f4ff221aedc
udioFlinger.cpp
udioFlinger.h
|
| b6333aa8317ce5162ab006c4baed6b0890936dc7 |
11-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Simplify ThreadBase::exit() aka requestExitAndWait"
|
| 3a144d08b713e3c0f6b7b8e95bc42cef5886f4fa |
11-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Disable HQ resamplers for now until qualified"
|
| 0d9302d7830b46542821b3e5f3e4f96942bd3cb3 |
11-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Move header declarations around for clarity"
|
| 858df80948ee64f478782a6a6c06533ba1651ef1 |
11-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Camel case readability & private disconnect(bool)"
|
| 95a87908ddf240760e8e8b35a6cafcc149c7f33f |
11-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove aliasing"
|
| c8ad36bbb30e99e49026cba78e5e0f83db5cb0f6 |
11-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use mul from audioutils"
|
| 31f0acff69efd46897221d847becca79ab15105f |
11-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Mark fields const if only set in constructor"
|
| b28686f95daee16edeb5f39af2cd5274ac3dc99f |
06-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Simplify ThreadBase::exit() aka requestExitAndWait
We can remove mExiting and use Thread::exitPending() instead.
The local sp<> on "this" in exit() is not needed, since the caller must
also hold an sp<> in order to be calling us. (Unless it was using a raw
pointer, but that would be dangerous for other reasons.)
Add comment explaining the mLock in exit().
Change-Id: I319e5107533a1a7cdbd13c292685f3e2be60f6c4
udioFlinger.cpp
udioFlinger.h
|
| 9d1f02d74fd395ec4de6861147da289423f0ab6f |
09-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Follow raw pointer and sp<> conventions
Unconditional delete for raw pointers.
Use "if (sp != 0)" not "if (sp.get() != 0)" or "if (sp != NULL)".
Use "if (raw != NULL)" not "if (raw)".
Change-Id: I531a8da7c37149261ed2f34b862ec4896a4b785b
udioPolicyService.cpp
|
| 0ba18ec1b343a8de70924f87630dd1f329b00fe6 |
10-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "No newline or space at end of ALOG format string"
|
| 12018d80add66f5558675614d73fa6549150806e |
10-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Move declaration of stream_type_t up earlier"
|
| 8b5980798ca06e57b1284e6e23fa220e1207bf41 |
10-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix typos in ALOG for pid vs tid"
|
| cdf2158f3b9498d6cd0eb228d8bee16e32399e16 |
02-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Disable HQ resamplers for now until qualified
This saves about 6500 bytes.
Change-Id: I87102fe561c95c19c9e615dea3de914f96639257
ndroid.mk
udioResampler.cpp
|
| 2f732eb768004c6362fae8a02c60b69c9400b032 |
26-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Move header declarations around for clarity
Put IAudioFlinger methods in binder opcode order.
Move hardware call state closer to where it is used.
getMode() and btNrecIsOff() are private.
Change-Id: Ie50340b396c39c763f2b155cbc08da8a0d0f2424
udioFlinger.h
|
| 004f719467c498942c40de9f260be601ee45e630 |
30-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Mark fields const if only set in constructor
Change-Id: Iacd06bb9efaf708cf965033be1f2297b58f7f75c
udioResampler.h
|
| d198b61603d5fa9298edea4ddb5852ea45159906 |
02-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Remove aliasing
Code was aliasing mBuffer as buffer, but continuing to use both buffer
and mBuffer after that point. This was at best misleading, and at worst
could confuse the compiler into generating bad code. There was no
performance advantage to the alias, in fact removing it saves 16 bytes.
Change-Id: I55023ddba465d9be82f66745b088d18af658ac60
udioResamplerSinc.cpp
|
| 58123c3a8b5f34f9d1f70264a3c568ed90288501 |
03-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Camel case readability & private disconnect(bool)
Change-Id: If66516ed2703e048c5e6ccc6cd431446a024f4a1
udioFlinger.cpp
udioFlinger.h
|
| 5b9ff43995f6a6b819d9ad37dd8cdc5ad4a088d7 |
09-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Use mul from audioutils
I verified that the disassembled output is identical.
Change-Id: I34a76f0842ebc4aef2c923e079e38d0bc1f98b5c
udioFlinger.cpp
|
| 23d82a9bc9a43b49ba684ba40875b91db310d3b9 |
03-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Fix typos in ALOG for pid vs tid
Change-Id: I6dc70f137d0ff8a86427ab8882a81886e1de0782
udioFlinger.cpp
udioPolicyService.cpp
|
| 90bebef5669a9385c706b042d146a31dca2e5d9b |
28-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
No newline or space at end of ALOG format string
Change-Id: I0bef580cbc818cb7c87aea23919d26f1446cec32
udioFlinger.cpp
udioResampler.cpp
udioResamplerCubic.cpp
|
| 6637baae4244aec731c4014da72418d330636ae1 |
09-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Fix races related to volume and mute
Fix race conditions when setting master volume, master mute, stream
volume, stream mute for a playback thread, and when reading stream
volume of a playback thread. Lock order is AudioFlinger, then thread.
Rename streamVolumeInternal to streamVolume_l, comment, and use it to
implement streamVolume().
Code size reduction:
- Remove dead code: AudioFlinger::PlaybackThread::masterVolume, masterMute, streamMute.
- Change return type of non-binder methods that always succeed from status_t to void.
- Remove virtual from volume and mute methods that don't need it.
This change saves 228 bytes but decreases performance of binder operations
due to the added locks.
Change-Id: Iac75abc1f54784873a667d1981b2e08f8f31e5c9
udioFlinger.cpp
udioFlinger.h
|
| b7bf796b758e144f94f6ed4d16c21bf22a118fb3 |
08-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Move declaration of stream_type_t up earlier
stream_type_t is used by AudioFlinger class, so it should be declared there.
This way we don't have to peek into PlaybackThread to get the declaration.
Change-Id: Ie08bab1604699214d1e8df2d48d3fbfbbc436e96
udioFlinger.h
|
| 02bbd20cece1785c223ac4ca2ddc635931a80673 |
08-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Rename type() to streamType()
This avoids possible confusion with thread's type().
Also remove redundant cast "(audio_stream_type_t)".
Change-Id: I320b9177b6c267a102d215f002228bcf988c437a
udioFlinger.cpp
udioFlinger.h
|
| 98ec94c5854daccc3474758524e7f4adfe535ce0 |
25-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Combine duplicate code & document wp<> in mClients
Change-Id: Iea8cfe8e57563337fb2484a1246ef79d6ad3db18
udioFlinger.cpp
udioFlinger.h
|
| 72ef00de10fa95bfcb948ed88ab9b7a177ed0b48 |
17-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_io_handle_t consistently instead of int
Other:
- add a comment to nextUniqueId
- made ThreadBase::mId const, since it is only assigned in constructor.
Change-Id: I4e8b7bec4e45badcde6274d574b8a9aabd046837
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
udioPolicyService.h
|
| dbfafaffe2e97eaf8d74ec6b6c468418a1ad2443 |
26-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Simplify destructors
Remove explicit clear() when the order doesn't matter.
Change-Id: I5931bc7ef5f681c7ce329aa9ec0a6e46d34a56c5
udioFlinger.cpp
|
| 5e92a7861196ddae14638d4b7a63fc4892b7ef59 |
30-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Effect UUID inputs passed by pointer are const
Change-Id: I1f5c338bcb7368e3dd8cd5f804b2e6d9fbe087f8
udioFlinger.cpp
udioFlinger.h
|
| 0a20fa9c41c96e31fa20e071074a4b6e7f6c41c3 |
08-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use pid_t not int"
|
| b61ec89bb0c701b3bd06eb658f854230681f8b39 |
08-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Don't double destruct audio_track_cblk_t"
|
| 63d2daed17ab749baa80bc808fb5083b688b771b |
08-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioFlinger methods const and inline"
|
| 86feafe15b3f9609e1e9f64184688c2b6f2e4834 |
08-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use virtual destructors"
|
| e616d4e6de6d53ddebbc3d7fb381af94589c2232 |
08-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Improve performance for sp<> on stack"
|
| 1579d7948117e3e6541b0cfda02cc5234a3280ea |
08-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use 0 not NULL for sp<> and wp<>"
|
| b5a66e15f371143f48665bbd462d75309582845d |
08-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use bool instead of int"
|
| e98bbd36d67243fe987b09904956550a68af1cc7 |
08-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Declare more IAudioFlinger methods const"
|
| 1a0ae5be3d1273cba12584b33830d859510fbf82 |
03-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Don't double destruct audio_track_cblk_t
Fortunately audio_track_cblk_t doesn't have a destructor, but for clarity
remove the double destruction.
Also add warning not to add any virtuals to audio_track_cblk_t.
Change-Id: I70ebe1a70460c7002145b2cdf10f9f137396e6f3
udioFlinger.cpp
|
| bb001926447d0b7dc71ca8bb3c9856f3136d8f4c |
03-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Use pid_t not int
Change-Id: Iad1c2fd4152e94080ad8c65c13ddf4519fc2ed27
udioFlinger.cpp
|
| d5e54f7a36daedc3b2a642d1499c262da04e6280 |
26-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Remove dead code
mFormat is unused in resampler
mClientTid is unused
local variable pid is unused in dump
Change-Id: Ib156e38029366620bfeff2a13e73471867155a5b
udioFlinger.cpp
udioFlinger.h
udioResampler.cpp
udioResampler.h
|
| f587ba5b991c7cd91e4df093d0d796bd419e5d67 |
27-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Declare more IAudioFlinger methods const
This is just documentation, as C++ method const-ness doesn't mean anything
for a binder API. Instead, here const means "no side effects".
Change-Id: Iaa9cd2fe477db10ae9a40cac4f79f0faa9b4e5e6
udioFlinger.cpp
udioFlinger.h
|
| 4c340c6521b634f159d2d6bc4e9359226fd8edf8 |
27-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use bool instead of int
The .h is not modified to avoid increasing data size.
Change-Id: Ide4a821a5b16424ffa03471dfff98dc3e9b5f751
udioMixer.cpp
|
| c59c004a3a6042c0990d71179f88eee2ce781e3c |
02-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
AudioFlinger methods const and inline
This saves 1063 bytes and probably improves performance.
Change-Id: I11cf0dfd925fbaec75e3d1b806852a538eae5518
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
udioMixer.h
udioResampler.h
|
| c19e22450e6e3d07594c935c7a9522e85e909e82 |
30-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use virtual destructors
It turns out to be just a comment, as all except AudioMixer are RefBase.
There are only a few performance-sensitive cases where it's worth thinking
about whether you need a virtual destructor, and the headache usually
outweighs the benefit.
Change-Id: I716292f9556ec17c29ce8c76ac8ae602cb496533
udioFlinger.h
udioMixer.h
udioResamplerSinc.h
|
| 7378ca506e4e20c2b2d4e94a131cf1b95831adb5 |
20-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use 0 not NULL for sp<> and wp<>
Change-Id: Id1f0c89acefaceed6cb9ca7c165fce895e46d85b
udioFlinger.cpp
udioPolicyService.cpp
|
| 787bae0578fbaab6219ebf23494866b224d01438 |
03-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use audio_in_acoustics_t consistently"
|
| ff3c83e04ae459c4252b86dd4440b4e37ee92c16 |
03-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use ToneGenerator::tone_type consistently"
|
| a0d68338a88c2ddb4502f95017b546d603ef1ec7 |
28-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use NULL not 0 for raw pointers
Use if (p != NULL) instead of if (ptr)
Change-Id: Iaec3413a59ccbf233c98fcd918cc7d70ac5da9fa
udioFlinger.cpp
udioMixer.cpp
udioPolicyService.cpp
|
| 87f155d6655b2d3b27e69281a29e85c6407e4d26 |
03-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "For performance, return large objects by reference"
|
| f81e97e4ec8b01965a5b36987f886cf5001f71ff |
03-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "No need to check a wp<> for 0 before promote()"
|
| 84afa3b51ac48f84ed62489529ce78cba7fca00e |
26-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Constructor initialization and const fields
In constructors, initialize member fields in the initialization list
rather than constructor body where possible. This allows more fields
to be const, provided they are never modified.
Also initialize POD fields in constructor, unless it's obvious they
don't need to be initialized. In that case, put a comment instead.
Remove explicit clear() in destructors on fields that are now const.
Give AudioSessionRef a default constructor, so it's immutable fields can
be marked const.
Add comment about ~TrackBase() trick.
Initialize fields in declaration order to make it easier to confirm that
all fields are set.
Move initialization of mHardwareStatus from onFirstRef() to constructor.
Use NULL not 0 to initialize raw pointers in initialization list.
Rename field mClient to mAudioFlingerClient, and getter from client()
to audioFlingerClient().
Change-Id: Ib36cf6ed32f3cd19003f40a5d84046eb4c122052
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
|
| 5c0ad10b14ec2287f90f95912d98e66eef006e2a |
03-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Cleanup thread types"
|
| 62da7fbd60bee2dd57f503126266e9f04311d400 |
03-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Make AudioTrack control block volume field private"
|
| d45ee9d9d61af0791c7c3c51f8d4fe6794ef02a5 |
02-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix const sp<>& in parameter list and return value"
|
| d05397144be774f2f3623c754e865f51753e4e31 |
30-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
For performance, return large objects by reference
Change-Id: Ibf737018ef1d3c7d717584615dcb2d4ecdb50c99
udioFlinger.cpp
udioFlinger.h
|
| 090f01963e215f895020a31e22368cd44e086ce3 |
30-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Improve performance for sp<> on stack
Combine default constructor for sp<> immediately followed by assignment,
as the reference-counting is relatively slow. Also return sp<> directly
rather than via local variable, for the same reason.
Change-Id: If55931f1e407994f6591ddde41b53db72fb4fc40
udioFlinger.cpp
|
| 435dbe6c3ecd04bcb4bd80584064e287ebccd720 |
30-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Fix const sp<>& in parameter list and return value
EffectModule::addHandle and Client::heap() were declared incorrectly.
As a parameter, an sp<> should be & for efficiency, and for input
parameters it should also be const to protect the caller's value.
But as a return value, an sp<> should have neither const or &. The "e"
in "return e;" might be located on the stack, and if there is "&" then
the caller would see the address of a variable which no longer exists.
Also, an & would make it hard to do "return 0;".
A "const" without & is meaningless in the return type.
(In this particular case, the "e" is a member field, so it was safe.)
Change-Id: I3df5f294214eb15a9d4d596c6d5ef29de97b5c27
udioFlinger.cpp
udioFlinger.h
|
| e9dd0176933d6233916c84e18f3e8c0d644ca05d |
28-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Unconditional delete
Don't check that pointer is non-NULL before delete.
Don't leave deleted member fields non-NULL, except in a destructor,
since it could be misleading in a dump or debugger. (mRsmpOutBuffer)
Change-Id: Ic0492a6b752f74a67f4c96dfb89ca2de4e69eecf
udioFlinger.cpp
udioMixer.cpp
udioPolicyService.cpp
|
| 77c1119ea0b5cb32287088ceeeb7e3b6bd14a85d |
25-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
No need to check a wp<> for 0 before promote()
Also remove unnecessary wp<> local variable.
Change-Id: I620e67b5d559d28616f8e00609a525cfe19c5ddc
udioFlinger.cpp
|
| 3d2f877c1cb4e4ae4ddde7f57f4353de9341f11b |
28-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use ToneGenerator::tone_type consistently
Also remove defaults in startToneCommand(), they're not needed and the
default for tone type was nonsense.
Change-Id: I70fa8cee4f3dbb8c66ceb3719c8d3d2f447f05b9
udioPolicyService.cpp
udioPolicyService.h
|
| de9719b3ec71472e6bf75117152176af51d1a515 |
27-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_in_acoustics_t consistently
Change-Id: I0a9dd668fb2e57b1c3ece3190588194974b99062
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
udioPolicyService.h
|
| fd267d7cf640225bb57f0ed1af44efc153275f6d |
27-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use audio_source_t consistently"
|
| a3a2cd4072aaa2d93c91251a786eb7323f8d2c27 |
27-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioStreamIn and AudioStreamOut"
|
| 6f5980b75df837231365d238c1b0d6f386363fbb |
27-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Declare methods in binder opcode order"
|
| 28f52c84c22e129063a576e1269a39ae0cc0bfb3 |
27-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use enum effect_state consistently"
|
| 114c458f2b80a252ec627add1d5fda2093c79068 |
27-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use enum track_state consistently"
|
| eba51fb3a361f67a6a64d5a16eba6084fe27d60e |
23-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_source_t consistently
Was a mix of audio_source_t, uint8_t, and int.
Related fixes:
- fix comments in MediaRecorder.java
- AudioPolicyService server side was not checking source parameter at
all, so if the client wrapper was bypassed, invalid values could be
passed into audio HAL
- JNI android_media_AudioRecord_setup was checking source for positive
values, but not negative values. This test is redundant, since already
checked at Java and now checked by AudioPolicyService also, but might
as well make it correct.
Change-Id: Ie5e25d646dcd59a86d7985aa46cfcb4a1ba64a4a
udioPolicyService.cpp
udioPolicyService.h
|
| aed850d0d3b3c8cf3feaf1438076f33db2a60946 |
26-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
AudioStreamIn and AudioStreamOut
These are immutable, so make the fields const.
getOutput() and getInput() methods are now const.
Change-Id: I128246ebd56ea50b3e542be43f2aa1bcb55f1373
udioFlinger.cpp
udioFlinger.h
|
| 23bb8becff20449a9b1647d5a1a99b14c83f0cce |
26-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Cleanup thread types
Use type_t instead of int for thread types.
Initialize ThreadBase::mType in constructor and make it const.
Change-Id: I43d141388b9639e4783c30b97dbda5688bf7555f
udioFlinger.cpp
udioFlinger.h
|
| 90716c5728b37637b2d0a730a721bfc9fad299e0 |
26-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Declare methods in binder opcode order
This makes it easier to compare interface and implementation.
Change-Id: Ie060e43dec348902abcf40f5a610cec639d6d0d3
udioFlinger.cpp
udioFlinger.h
|
| 29c23c3aee5ae799b3480dc6876a46c46b019710 |
26-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use enum mixer_state consistently
Change-Id: I5b71ed20f939dfc4b98143334b7aa064d282f584
udioFlinger.cpp
udioFlinger.h
|
| 28243dd563fee1c82f0fff6cc27b5cbf21fa2585 |
26-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use enum effect_state consistently
Also fix indentation
Change-Id: I393ef9e37ffceed5ad4a78df439726ae1fe139df
udioFlinger.h
|
| b853e986caf43408ad95b9014f194aadff385e3c |
26-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use enum track_state consistently
Change-Id: Ie5ebb7befa092e1de1e4df9c6e2d51e6bcfd176a
udioFlinger.cpp
udioFlinger.h
|
| 9365ea9bf2e439b3e71abbabe22ce7382ebc4b3a |
25-Jan-2012 |
Eric Laurent <elaurent@google.com> |
am 535b0264: am 7eeaf3f0: Merge "AudioFlinger: refine mixer sleep time logic" into ics-mr1
* commit '535b0264a4cfa790e549bd9cd09980788f1375f4':
AudioFlinger: refine mixer sleep time logic
|
| 84e19873fde204d73628ba1b5ca9e3f5778574fa |
24-Jan-2012 |
Eric Laurent <elaurent@google.com> |
am 7eeaf3f0: Merge "AudioFlinger: refine mixer sleep time logic" into ics-mr1
* commit '7eeaf3f07aa6fb10639d9f96c1367eb98c3e8839':
AudioFlinger: refine mixer sleep time logic
|
| 7c5aea0a8d9b422999483f96a2566f77ff11abf2 |
24-Jan-2012 |
Eric Laurent <elaurent@google.com> |
am 41773d46: Merge "DO NOT MERGE Revert "Revert "AudioFlinger: mix track only when really ready (2)""" into ics-mr1
* commit '41773d46556aa47d4322ff89fdaf7d1345c2d1f2':
DO NOT MERGE Revert "Revert "AudioFlinger: mix track only when really ready (2)""
|
| 21e4b6ed00e814bffc70895847a4944d7a190020 |
24-Jan-2012 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: refine mixer sleep time logic
When an AudioTrack is in underrun state, the AudioFlinger mixer will
sleep for a short period of time to give the app a chance to fill the
AudioTrack buffer. If the AudioTrack is still not ready during next mixing round,
the mixer will proceed with other tracks.
If an application keeps a steady underrun condition, the AudioFlinger mixer will
alternate between ready and not ready states. In the longer term this will cause the
audio HAL to underrun.
There is a mechanism to reduce the sleep period if the mixer is not ready several times in a
row but this mechanism is defeated by the alternating ready/not ready conditions.
The fix consists in only increasing sleep time if the mixer is ready for at least two
consecutive times.
Issue 5904527.
Change-Id: Id0139bca9be8c4e425ec6d428515c4d8f718e8c9
udioFlinger.cpp
|
| eaa0b5cc2f7723e9b25298126d0dcb48c56d5dac |
24-Jan-2012 |
Eric Laurent <elaurent@google.com> |
DO NOT MERGE Revert "Revert "AudioFlinger: mix track only when really ready (2)""
This reverts commit b918035d34422a2041b6ec8c09c566bb93345b40.
Change-Id: I093bcfa56ad54a080b930208b6b79169d33581fb
udioFlinger.cpp
udioFlinger.h
|
| 15dfda272eec983508b89fb8bc9ca6f2bb825496 |
24-Jan-2012 |
Justin Ho <justinho@google.com> |
am fee5a860: Merge "DO NOT MERGE Revert "AudioFlinger: mix track only when really ready (2)"" into ics-mr1
* commit 'fee5a860a8355cda071ff23644e943414ba7f65d':
DO NOT MERGE Revert "AudioFlinger: mix track only when really ready (2)"
|
| 7baf7894bc2f1a62440f381eeb50143f210a5d61 |
23-Jan-2012 |
Justin Ho <justinho@google.com> |
DO NOT MERGE Revert "AudioFlinger: mix track only when really ready (2)"
This reverts commit 71c4496a9757438afd30b4404824f296f6158a49.
Change-Id: Iff10c49ea728bb10023ddeb50a3b708db770fff2
udioFlinger.cpp
udioFlinger.h
|
| 335787fe43596f38ea2fa50b24c54d0823a3fb1d |
21-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Remove AudioFlinger dependencies on client
Change-Id: Ibb591e41a3ca5d7015e2b66b98b8fef5f415fb37
udioFlinger.cpp
udioFlinger.h
|
| 83d86538c4c479a9225c75ab27938e8f05abb9c8 |
17-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Make AudioTrack control block volume field private
This is part of the process of abstracting the control block
to make it easier to maintain.
Change-Id: Idb8f461e68dab3bcf268159cc0781651c6fb7094
udioFlinger.cpp
|
| 58f30210ea540b6ce5aa6a46330cd3499483cb97 |
12-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_format_t consistently, continued
Was int or uint32_t.
When AudioFlinger::format can't determine the correct format,
return INVALID rather than DEFAULT.
Init mFormat to INVALID rather than DEFAULT in the constructor.
Subclass constructors will set mFormat to the correct value.
Change-Id: I9b62640aa107d24d2d27925f5563d0d7407d1b73
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
udioPolicyService.h
|
| d967f0a099db2b71597a3127134afd4a46287a4a |
20-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove redundant get()"
|
| 7fc9a6fdf146ded90b51c52f4a05d797294dcb85 |
10-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Remove redundant get()
get() is almost always unnecessary, except in a LOG.
Also no need to check for != 0 before calling get().
Change-Id: Ib06e7a503f86cf102f09acc1ffb2ad085025516d
udioFlinger.cpp
|
| ea3cc3bca949139e401b77f2ac0cce7ac6e76f8f |
20-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove dead setRingerMode(mode, mask)"
|
| a3db30f7c53936a3acb87d2c44807ee2f0cfa16e |
20-Jan-2012 |
Jean-Baptiste Queru <jbq@google.com> |
am 9d25b82d: Merge "Rename LOG_ASSERT to ALOG_ASSERT"
* commit '9d25b82d280c2d979d500e7da4447148f32f820b':
Rename LOG_ASSERT to ALOG_ASSERT
|
| d746737921074e2a6c39c52b06022c5166689df5 |
20-Jan-2012 |
Jean-Baptiste Queru <jbq@google.com> |
am 6df477be: Merge "Rename (IF_)LOGE(_IF) to (IF_)ALOGE(_IF)"
* commit '6df477be186233e36fc370c4d2db6c1ed928a740':
Rename (IF_)LOGE(_IF) to (IF_)ALOGE(_IF)
|
| daef36f5d4934bd055c694a8d54b86e2b50a6159 |
20-Jan-2012 |
Jean-Baptiste Queru <jbq@google.com> |
am a826f9e2: Merge "Rename (IF_)LOGW(_IF) to (IF_)ALOGW(_IF)"
* commit 'a826f9e2c4f6329d8d48c927f6e942e78ffaf92f':
Rename (IF_)LOGW(_IF) to (IF_)ALOGW(_IF)
|
| 3734cbca25c6d902677cfb5e59dff7a1cb17a45d |
20-Jan-2012 |
Jean-Baptiste Queru <jbq@google.com> |
am 4f367f33: Merge "Rename (IF_)LOGI(_IF) to (IF_)ALOGI(_IF)"
* commit '4f367f3387887c538c81c34cc8becaea6fa5e430':
Rename (IF_)LOGI(_IF) to (IF_)ALOGI(_IF)
|
| e744a90eb52bf9547848c08380cfb7ba7e63ffbc |
20-Jan-2012 |
Jean-Baptiste Queru <jbq@google.com> |
am ba7f0d2a: Merge "Rename (IF_)LOGD(_IF) to (IF_)ALOGD(_IF)"
* commit 'ba7f0d2a03643ce429421b81febf18fd50473070':
Rename (IF_)LOGD(_IF) to (IF_)ALOGD(_IF)
|
| ca05a87f4c69670a06bdee4b4f98bcdcd838beda |
20-Jan-2012 |
Jean-Baptiste Queru <jbq@google.com> |
Merge ee4618bc
Change-Id: Ie1dc6ad38e7c30636d80f6caef11cf6673144940
|
| 39fd8eb956dea32692d3d72b10b64a8597df0a40 |
09-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename LOG_ASSERT to ALOG_ASSERT
Change-Id: Ie2c7ea6560656d65bad791a61996174c75677517
udioResampler.cpp
udioResamplerCubic.cpp
|
| 9a8ded7348c5b2302dd27b285b395416bc842c49 |
06-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGE(_IF) to (IF_)ALOGE(_IF)
Change-Id: I1de629b4632a4b3187ca1a28d6416daccd35f924
udioFlinger.cpp
udioMixer.cpp
udioPolicyService.cpp
udioResampler.cpp
|
| aa70226152d2084f85a96b52359dbc8476a86a45 |
06-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGW(_IF) to (IF_)ALOGW(_IF)
Change-Id: I8fbdfa7a7581f481968dbb65aa40f7042936d7cb
udioFlinger.cpp
udioMixer.cpp
udioPolicyService.cpp
udioResamplerCubic.cpp
|
| 53feeb42c721e8fc9285e35e679906a951f3277c |
04-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGI(_IF) to (IF_)ALOGI(_IF)
Change-Id: I26f76452ac49e2890b14d133c065493d8df0fb4a
udioFlinger.cpp
udioPolicyService.cpp
|
| 52546c0ef96aa3e7e21482e0f9b6e982557c8da9 |
20-Dec-2011 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGD(_IF) to (IF_)ALOGD(_IF)
Change-Id: I44f267700356967dc51e8f85ebf457dc85cfb229
udioFlinger.cpp
udioMixer.cpp
udioResampler.cpp
|
| 3812256de32e73e38ba16e50ac0451c10223d4eb |
20-Oct-2011 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGV(_IF) to (IF_)ALOGV(_IF)
Change-Id: I5321ebd12e9c6248a108529e82c4e1af2a4405e3
udioFlinger.cpp
udioMixer.cpp
udioPolicyService.cpp
udioResampler.cpp
|
| aeeb7e219e34d2d657d829913659a4e10e976375 |
19-Jan-2012 |
Eric Laurent <elaurent@google.com> |
resolved conflicts for merge of 05683c85 to master
Change-Id: I7846b7da8c5813b7a9b1f3f71aede0229689ff0d
|
| 86905f445c96743b037eed6360afa956a85b019c |
19-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Add units to time periods"
|
| ca42b286a162f5cd6a4a916bfe429abe95780730 |
19-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use const const strings for arrays and pointers to strings"
|
| 22ecc912a87099cff8cadc424cd12f85c118673f |
09-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Add units to time periods
Change-Id: Ib980e2676cecf5d9b0c2e388da6dc8e370df8abb
udioPolicyService.cpp
|
| 8dad0e31c2366c501bd1d600261d0af35a6ca786 |
09-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use const const strings for arrays and pointers to strings
Change-Id: I0a073e17ebf6ad59e7edbedac104c6fa4bcaf9c9
udioPolicyService.cpp
udioPolicyService.h
|
| f1d4592d4c3c99ebab55559e164ff102e825283e |
14-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
For booleans, use ! instead of == false
Change-Id: Ibc115936d2d0b0b7744ebe9b52839ea5b42c4edd
udioFlinger.cpp
udioPolicyService.cpp
|
| 0b07b8085d7b837b4dd5f09e0c8c39408f6bdbf7 |
18-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Remove dead setRingerMode(mode, mask)
Change-Id: Ia4cc8be8424a40b3dcb7ebd0264fdff4e5247f7f
udioPolicyService.cpp
udioPolicyService.h
|
| 2774144fa8283f1a7b43e17a53c97dec0c366dd3 |
18-Jan-2012 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: mix track only when really ready (2)
This problem due to the way audio buffers are mixed when
low power mode is active was addressed by commits 19ddf0eb
and 8a04fe03 but only partially. As a matter of fact, when more
than one audio track is playing, the problem is still present.
This is most noticeable when playing music with screen off
and a notification or navigation instruction is played: in this case,
the music or notification is likely to skip.
The fix consists in declaring the mixer ready if all active tracks
are ready. Previous behavior was to declare ready if at least one track was
ready. To avoid that one application failing to fill the track buffer blocks other
tracks indefinitely, this condition is respected only if the mixer was ready
in the previous round.
Issue 5799167.
Change-Id: Iabd4ca08d3d45f563d9824c8a03c2c68a43ae179
udioFlinger.cpp
udioFlinger.h
|
| 04eaf3e5f20d7faa956c0d672024ffb5117f4c26 |
18-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Track volume cleanup"
|
| 63ad6aacc6ce6b729bf25f41376cfea731a2c1eb |
18-Jan-2012 |
Eric Laurent <elaurent@google.com> |
Merge "audio framework: manage stream volume per device"
|
| b1cf75c4935001f61057989ee3cf27bbf09ecd9c |
17-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Track volume cleanup
Always read and write track volumes atomically. In most places this was
already being done, but there were a couple places where the left and
right channels were read independently.
Changed constant MAX_GAIN_INT to be a uint32_t instead of a float.
It is always used as a uint32_t in comparisons and assignments.
Use MAX_GAIN_INT in more places.
Now that volume is always accessed atomically, removed the union
and alias for uint16_t volume[2], and kept only volumeLR.
Removed volatile as it's meaningless.
In AudioFlinger, clamp the track volumes read from shared memory
before applying master and stream volume.
Change-Id: If65e2b27e5bc3db5bf75540479843041b58433f0
udioFlinger.cpp
|
| 83844cc2f95dc279015b47fd1e18c7cb4eabe9a1 |
19-Nov-2011 |
Eric Laurent <elaurent@google.com> |
audio framework: manage stream volume per device
Improve volume management by keeping track of volume for each type
of device independently.
Volume for each stream (MUSIC, RINGTONE, VOICE_CALL...) is now maintained
per device.
The main changes are:
- AudioService now keeps tracks of stream volumes per device:
volume indexes are kept in a HashMap < device , index>.
active device is queried from policy manager when a volume change request
is received
initalization, mute and unmute happen on all device simultaneously
- Settings: suffixes is added to volume keys to store each device
volume independently.
- AudioSystem/AudioPolicyService/AudioPolicyInterface: added a device argument
to setStreamVolumeIndex() and getStreamVolumeIndex() to address each
device independently.
- AudioPolicyManagerBase: keep track of stream volumes for each device
and apply volume according to current device selection.
Change-Id: I61ef1c45caadca04d16363bca4140e0f81901b3f
udioPolicyService.cpp
udioPolicyService.h
|
| ad0f6cc5e115ca167ff122c83451b46d85c590ac |
17-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Remove dead setVolume() and mVolume[2]
Change-Id: I94b835434093e920432614eb5007101e87758f32
udioFlinger.cpp
udioFlinger.h
|
| 0696400a6bb9abbed62b3b9c6aa105495dc600a2 |
17-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use audio_mode_t consistently"
|
| 263709e7be37c7040aaef385bc5c9389a9b5f514 |
06-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Check stream type in AudioFlinger::createTrack
A bad parameter to AudioFlinger::createTrack could cause mediaserver to crash.
Other AudioFlinger stream type cleanup:
- Simplify range check for audio_stream_type_t
- Add comment about mStreamTypes array initialization.
Change-Id: Ia33aa1cce0fdd694b08d9288816ffc097a9543d0
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
|
| 3944e0326a286bcb931551e61e79c033b10d09d4 |
17-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix locking for mMasterVolume and mMute"
|
| 613882293184e575a44bff681a3decaefe889e69 |
17-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use size_t for frame size"
|
| 0107954f72153db747a3727dc1157e9236dfed90 |
17-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use audio_stream_type_t consistently"
|
| 9806710f5d6722cfc5783c7eca3512451a0f2035 |
13-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Fix locking for mMasterVolume and mMute
mMasterVolume and mMute are both protected by mutex in AudioFlinger class, but
there were two places where they were accessed without a mutex.
Also make AudioFlinger::mMasterMute private not protected.
Change-Id: Ia3897daeb5c50313df5bcc071824357526237f3e
udioFlinger.cpp
udioFlinger.h
|
| 05632a5fa4b88ca474294887fc92a9fcdf0e2352 |
03-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
AudioTrack and AudioFlinger send level cleanup
Add an API to control block for getting/setting send level.
This allow us to make the mSendLevel field private.
Document the lack of barriers.
Use 0.0f to initialize floating-point values (for doc only).
Change-Id: I59f83b00adeb89eeee227e7648625d9a835be7a4
udioFlinger.cpp
|
| b9980659501d0428d65d8292f3c32da69d37fbd2 |
11-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use size_t for frame size
except in the control block, where we don't have room.
In AudioFlinger::ThreadBase::TrackBase::getBuffer,
read the frame size from control block only once.
Change-Id: Id6c4bccd4ed3e07d91df6bbea43bae45524f9f4e
udioFlinger.cpp
udioFlinger.h
|
| fff6d715a8db0daf08a50634f242c40268de3d49 |
13-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_stream_type_t consistently
At native level it was a mixture of audio_stream_type_t, int, uint32_t,
and uint8_t. Java is still int. Also fixed a couple of hard-coded -1
instead of AUDIO_STREAM_DEFAULT, and in startToneCommand a hard-coded 0
instead of AUDIO_STREAM_VOICE_CALL.
Change-Id: Ia33bfd70edca8c2daec9052984b369cd8eee2a83
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
udioPolicyService.h
|
| f78aee70d15daf4690de7e7b4983ee68b0d1381d |
04-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_mode_t consistently
It was int or uint32_t.
Also make getMode() const.
Change-Id: Ibe45aadbf413b9158e4dd17f2b3bcc6355288d37
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
udioPolicyService.h
|
| e3a067f8bc98134941ee1a4da8c2a92a15aaa9cc |
11-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Simplify range check for audio_mode_t"
|
| c40256146bee58bff09e1c16ef99ea06d31f89f9 |
11-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use correct type for hardware call state"
|
| 2ea3410d0d3d592ce30c3ba0ce3e0e63b1244057 |
11-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use consistent style of & reference for AutoMutex"
|
| 9770988e61961d34033fd2c12f0de85a267df68f |
11-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioFlinger new can't fail"
|
| 02b2d2dddc9be339cf7647d4ead53cecd1c51038 |
11-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix build warning"
|
| 930f4caa1e311ef7ff538c421a324396157eb24f |
07-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Simplify range check for audio_mode_t
AudioSystem::setMode previously allowed negative modes, but these were
then rejected by AudioFlinger.
Now negative modes (including AUDIO_MODE_INVALID and AUDIO_MODE_CURRENT)
are explicitly disallowed.
Change-Id: I0bac8fea737c8eb1f5b6afbb893e48739f88d745
udioFlinger.cpp
udioPolicyService.cpp
|
| c1dc1cb1d1eaf84e88669f1a5f22579a0d9237c2 |
09-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename LOG_ASSERT to ALOG_ASSERT DO NOT MERGE
See https://android-git.corp.google.com/g/157519
Bug: 5449033
Change-Id: I8ceb2dba1b031a0fd68d15d146960d9ced62bbf3
udioFlinger.cpp
udioResampler.cpp
udioResamplerCubic.cpp
|
| 5f29ca38b71506ad7c7cb9925efbddf588e9655b |
09-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "By convention const goes before the type specifier"
|
| 25b248eb52a0a16adaef6b79c9d92cb88b9a2bc2 |
04-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use cached reference to media.player service
This save unnecessary binder calls
Change-Id: I93a60efc54d9c8fb8fab706cd4477bbfd00ffec8
udioFlinger.cpp
|
| 879bc64e43d62f2690c77a3282cef2e45d1b76e9 |
09-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Replace loop by __builtin_ctz"
|
| febdbfec3b1ed0e20aa4f10bfdd82702d3e41f4b |
09-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "suspended() and isSuspended() are const"
|
| 29357bc2c0dd7c43ad3bd0c8e3efa4e6fd9bfd47 |
06-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGE(_IF) to (IF_)ALOGE(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/#/c/157220
Bug: 5449033
Change-Id: Ic9c19d30693bd56755f55906127cd6bd7126096c
udioFlinger.cpp
udioMixer.cpp
udioPolicyService.cpp
udioResampler.cpp
|
| 4a6f02833d8421b7d9b20f89729d8bb03b8b8102 |
07-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
AudioFlinger new can't fail
Change-Id: I7dae05a5ea1c962a9975386eab1fedbbe106ffba
udioFlinger.cpp
|
| a7d8d6fc5e102a08b262a1b78aa1abeeb097d9e4 |
06-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use consistent style of & reference for AutoMutex
AutoMutex, which is a typedef for Mutex::Autolock, is overloaded for
either a reference (&) or pointer (*) parameter, but we prefer to use
the reference form when the mutex is known at compile time.
Change-Id: I3515e6d6ab7959b2356a27fa3b04fd49e42cb31e
udioFlinger.cpp
|
| a4454b4765c5905f14186893b0688be375642283 |
04-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use correct type for hardware call state
Change-Id: Ic6d98b129e3ec653df1d8f7e829adf8dccb4f378
udioFlinger.cpp
udioFlinger.h
|
| 54c3b66444ebfb9f2265ee70ac3b76ccefa0506a |
06-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
By convention const goes before the type specifier
Change-Id: I70203abd6a6f54e5bd9f1412800cc01212157e58
udioFlinger.h
udioMixer.cpp
udioMixer.h
udioResamplerSinc.cpp
udioResamplerSinc.h
|
| f9a27779634ce3a01e5957f234cd04eba74fa07f |
06-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Fix build warning
Change-Id: Ic99608d0c14ed56c02f036e0bbaaae1b16bab8ba
udioMixer.cpp
|
| 5ff1dd576bb93c45b44088a51544a18fc43ebf58 |
06-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGW(_IF) to (IF_)ALOGW(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/157065
Bug: 5449033
Change-Id: I00a4b904f9449e6f93b7fd35eac28640d7929e69
udioFlinger.cpp
udioMixer.cpp
udioPolicyService.cpp
udioResamplerCubic.cpp
|
| 98dd542ffcd3424b1da58bd273c0c1ff5b8c8960 |
15-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Replace loop by __builtin_ctz
Using the builtin is faster on some platforms, for example on ARM it's
19 instructions instead of 13, and is O(1) instead of O(n). Of course,
track creation is an inherently slow operation, so this doesn't matter
much now. But if we add support for virtual tracks, then physical tracks
will be allocated/freed more frequently. Also just on principle ...
Change-Id: I3f590934092bd7a1869cbedbc7357928aa5cc8ff
udioMixer.cpp
|
| 8a08dcc0a5de19a904e77d5f31bed3dff9a59890 |
05-Jan-2012 |
Steve Block <steveblock@google.com> |
Merge "Rename (IF_)LOGI(_IF) to (IF_)ALOGI(_IF) DO NOT MERGE"
|
| a3a854868a80fd9b9b8720e06a172754943f9417 |
04-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
suspended() and isSuspended() are const
Change-Id: I04b95970b5a645b64e7e64fffd46d868354dda66
udioFlinger.cpp
udioFlinger.h
|
| 88592eccaf6afcddd5f985955be92fe25205c680 |
05-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove the notion of "active track" from mixer"
|
| a2a0a5d7d56baa831870f4bf2a0d942a477d92ef |
05-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use the standard CC_LIKELY and CC_UNLIKELY macros"
|
| f6b1678f8f508b447155a81b44e214475ab634a8 |
15-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Use the standard CC_LIKELY and CC_UNLIKELY macros
Several source files privately defined macros LIKELY and UNLIKELY in terms
of __builtin_expect. But <cutils/compiler.h> already has CC_LIKELY and
CC_UNLIKELY which are intended for this purpose. So rename the private
uses to use the standard names.
In addition, AudioFlinger was relying on the macro expanding to extra ( ).
Change-Id: I2494e087a0c0cac0ac998335f5e9c8ad02955873
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
udioMixer.h
|
| 7ab41c9f773ba599646f1b0d00955c1be80f92fd |
05-Jan-2012 |
Eric Laurent <elaurent@google.com> |
resolved conflicts for merge of 1a4b9939 to master
Change-Id: I0c910d391a38a916d8431f7d1f5b82e39e1a66c2
|
| df64d15042bbd5e0e4933ac49bf3c177dd94752c |
04-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGI(_IF) to (IF_)ALOGI(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/156801
Bug: 5449033
Change-Id: Ib08fe86d23db91ee153e9f91a99a35c42b9208ea
udioFlinger.cpp
udioPolicyService.cpp
|
| b8a805261bf0282e992d3608035e47d05a898710 |
20-Dec-2011 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGD(_IF) to (IF_)ALOGD(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/156016
Bug: 5449033
Change-Id: I4c4e33bb9df3e39e11cd985e193e6fbab4635298
udioFlinger.cpp
udioMixer.cpp
udioResampler.cpp
|
| 071ccd5a9702500f3f7d62ef881300914926184d |
23-Dec-2011 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix clicks on 48kHz audio.
The calculation done in prepareTracks_l() for the minimum amount
off frames needed to mix one output buffer had 2 issues:
- the additional sample needed for interpolation was not included
- the fact that the resampler does not acknowledge the frames consumed
immediately after each mixing round but only once all frames requested have been used
was not taken into account.
Thus the number of frames available in track buffer could be considered sufficient although
it was not and the resampler would abort producing a short silence perceived as a click.
Issue 5727099.
Change-Id: I7419847a7474c7d9f9170bedd0a636132262142c
udioFlinger.cpp
udioMixer.cpp
udioMixer.h
udioResampler.h
|
| 9c56d4ae6212c21ce5fd71ed534eb195983a07c1 |
20-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Remove the notion of "active track" from mixer
This is a first step towards making the mixer more object-oriented.
Change-Id: Ifd445d0e471023a7f5c82e934736ffc95ba1b05b
udioFlinger.cpp
udioMixer.cpp
udioMixer.h
|
| 12d9702773c8a3242a44dd6528126c9ff0dade43 |
20-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove dead code"
|
| bbaf8673f1d1dd79d1b7f474ca7111da58e84aff |
20-Dec-2011 |
Eric Laurent <elaurent@google.com> |
Merge "audio effects: rename configure command"
|
| 3d5188bd6abe55898f10a0edf3c05aff8aa2ef67 |
17-Dec-2011 |
Eric Laurent <elaurent@google.com> |
audio effects: rename configure command
Renamed audio effect library interface command for audio format
configuration from EFFECT_CMD_CONFIGURE to EFFECT_CMD_SET_CONFIG.
This makes the naming more consistent with other exixsting commands
and allow adding a new command to get the configuration (EFFECT_CMD_GET_CONFIG).
Same change for reverse channel configuration renamed from
EFFECT_CMD_CONFIGURE_REVERSE to EFFECT_CMD_SET_CONFIG_REVERSE.
Implemented EFFECT_CMD_GET_CONFIG in exisitng effect libraries.
Change-Id: Ia7b1c620f13797fe5aceb3b0b4acbacce09fb067
udioFlinger.cpp
|
| 81a028fef62bcadf13fc8550067a3d29c918b3ca |
15-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Remove dead code
Change-Id: Icf23f7f90fdeb660f4015f22cf239e6d05f5d03c
udioMixer.cpp
udioMixer.h
|
| f602a5f7f29e694a7da4c143737b74501c00477d |
17-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Use constants for 2 and 32"
|
| d1e672acd8fa1af899f85ee2321327237028adf8 |
17-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Extract out audio DSP code to utility library"
|
| bf71f1e7948406492376c6cbd5e6a30c8cb670e4 |
13-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Use constants for 2 and 32
Change-Id: If820dfd58b6df258570750610a07af99598d9e53
udioMixer.cpp
udioMixer.h
|
| b92bc47a05925c29efef0d66711b6d3f17f406e7 |
16-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Use switch in AudioMixer::setParameter"
|
| 788040c5189bbdaf567ce4b29ffd1db08ea1020c |
05-May-2011 |
Glenn Kasten <gkasten@google.com> |
Use switch in AudioMixer::setParameter
Replace series of if/then/elses by easier-to-read switch. Also return
void instead of status_t, since callers weren't checking it. Assert on
bad input parameters.
Change-Id: Ie1f0a297977b28501d20e1af819afed9b4750616
udioMixer.cpp
udioMixer.h
|
| 237a624f674800d2300806b115eee8c9bb7db033 |
16-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
deleteTrackName now asserts on bad input parameter
This is safe, as the input parameter is always track->name(),
which must be valid.
Change-Id: Iea8ea3a5706c27026335526ba8851030d00681f8
udioMixer.cpp
|
| 3b21c50ef95fe4e7ac3426ca14b365749e66ff08 |
15-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Extract out audio DSP code to utility library
Change-Id: Ib8ce72028a7ea30e82baa518e381370e820ebbd0
ndroid.mk
udioFlinger.cpp
udioMixer.cpp
udioMixer.h
|
| 6d4bf6eda04888144ec59cba4816a2cbb79b9e5c |
16-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "setActiveTrack and setBufferProvider can't fail"
|
| b87c068727a15a3d3f0bfdcb758c76a097f5e869 |
16-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Simplify enable/disable mixing"
|
| 079123ee3d2e20bbc17a7ddbd96ca46bed27898f |
16-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Improve resistance to leaks for ConfigEvent"
|
| 26fa039c3752eaaf74d1be53d9795f48e9f43de3 |
16-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Use NULL not 0 for pointers"
|
| fba380a0b2c62684ce4272a05165965bae29d6a1 |
16-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
setActiveTrack and setBufferProvider can't fail
Return void, not status_t, from setActiveTrack and setBufferProvider.
These methods returned status_t, but the callers never checked the
return value. Since these aren't externally visible APIs, they now
return void, and assert on bad input parameters.
Change-Id: I530ed29484596ae41e8659826ca425149c51c2a1
udioMixer.cpp
udioMixer.h
|
| 1c48c3c61970527b97892ab6a2daae8eaac26964 |
15-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Simplify enable/disable mixing
The MIXING enum isn't needed, and now returns void instead of status_t.
Change-Id: Ibe4ec24081d75ad4ab78b9c7191fc9077959c4e9
udioFlinger.cpp
udioMixer.cpp
udioMixer.h
|
| f3990f2cc8fd824ae52a880a7b22248e1bdfb192 |
13-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Improve resistance to leaks for ConfigEvent
A Vector of pointers is risky, as there is no ownership (and the
ThreadBase destructor was not deleting them, so if there were any left
over at end it would leak). Replaced by a Vector of values.
Change-Id: Iddde72dc30134adfcf724dec26cbe0a742509b8c
udioFlinger.cpp
udioFlinger.h
|
| e0feee3da22beeffbd9357540e265f13b2119cbb |
13-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Use NULL not 0 for pointers
Change-Id: Iab3f9abbdab617dc5a599e657ec46a0b0a002eef
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
|
| 42968939dfce0954d6540011199045ec4ed7de80 |
15-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix indentation and whitespace"
|
| a06a9a50b37d60e9c43c9de9f8ea3a8649cd5691 |
15-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Improve AudioFlinger error logging"
|
| 53a166cabc956529d2336b04c26b9c03b550ab38 |
15-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioMixer uninitialized simple fields"
|
| 0cfd8231e4c489392809bf44c174315df2690145 |
13-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
AudioMixer uninitialized simple fields
Change-Id: I47e3849f048d4d990c9634f9c8a2ca2de0339e93
udioMixer.cpp
|
| d879601ace079e3c0aed79cf3fa5fb4db6ad4a9f |
28-Oct-2011 |
Glenn Kasten <gkasten@google.com> |
Improve AudioFlinger error logging
Change-Id: I8ce9aff4038cd7fa0067600faa8080b137db1939
udioFlinger.cpp
|
| 2eda60a8485cfe70a60e72156beffdc470ecb093 |
15-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Audio C++ comments"
|
| c5ac4cb3a5124860ccfc7e4ff66251c55a5595ca |
12-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Fix indentation and whitespace
Use git diff -w to verify.
Change-Id: Ib65be0a1ecf65d6cad516110604e3855bf68a638
udioFlinger.cpp
udioMixer.cpp
|
| c23bd9b5b9e4be9c395789810fdd8522296fc50c |
15-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Use const char correctly in AudioFlinger"
|
| e5dfcd8c6792c4b64120fd03708729b70a887f2a |
15-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Use units after all times"
|
| b299dc4ded29a226daac07f195d1558e660d2f9f |
15-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove redundant clear()"
|
| 362c4e697d8e9c034e964ac7b40227e054491547 |
14-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Audio C++ comments
Change-Id: I84906ebb9dfcfa5b96b287d18364b407f02a30c1
udioFlinger.cpp
udioMixer.h
|
| d1d8f23dde239289ba8c0de0944baaca3586c880 |
08-Jul-2011 |
Glenn Kasten <gkasten@google.com> |
Fix includes
Remove unused #include dlfcn.h
Add #include String8.h
Forward class declaration is only for references and pointers.
In this case, we need the full class declaration,
and were incorrectly depending on another header file to do it.
Change-Id: Iff65cceb3982698cc1ebaeec405695fc190c03ff
udioPolicyService.cpp
udioPolicyService.h
|
| 91eb8bfbe253a6b6fe1aa23fb884a601c28991c4 |
13-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Remove redundant clear()
Change-Id: Ie5e4e63cbc8fa85ef50451dddf8f149fa864b132
udioFlinger.cpp
|
| ec1d6b5e17281a066d618f7fcd2b63b3ce11f421 |
12-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Use const char correctly in AudioFlinger
Use const char [] instead of const char * to eliminate unnecessary pointer.
Make the array audio_interfaces also const, in addition to the strings.
Change-Id: I31f33d1dcb9a657ee136f4280fd2d46492496831
udioFlinger.cpp
|
| 7dede876998ff56351d495ec3a798c1b131193e8 |
13-Dec-2011 |
Glenn Kasten <gkasten@google.com> |
Use units after all times
Change-Id: I48d3f29c37228b5d03189e4c9600824c9360cac9
udioFlinger.cpp
|
| 2013d4d159bfc29b4143d3b5fd4735f51a03684c |
06-Dec-2011 |
Eric Laurent <elaurent@google.com> |
am 5433e25f: am 7b6aff23: Merge "audioflinger: fix audio skipping over A2DP" into ics-mr1
* commit '5433e25f6ce013860ff2a074ad8d1158cc39ab91':
audioflinger: fix audio skipping over A2DP
|
| 162b40bbaf3c3a24f61a6636bef6f80a9c0a31dd |
05-Dec-2011 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix audio skipping over A2DP
The maximum sleep time allowed in the mixer thread when audio tracks
are enabled but not ready for mixing is derived from the latency
reported by the output stream.
This does not work for A2DP where the latency also reflects encoding, decoding
and transfer time.
Modified activeSleepTimeUs() to take A2DP case into account.
Issue 5682206.
Change-Id: I3784ac01fb6f836b5a6ce6f764fb15347586de35
udioFlinger.cpp
udioFlinger.h
|
| 926798f8c21ab002d9797ef8973852a2612c1f75 |
23-Nov-2011 |
Eric Laurent <elaurent@google.com> |
am f6422f5f: am 20398fac: Merge "audioflinger: reduce sleep time to avoid underrun" into ics-mr1
* commit 'f6422f5f5d04aab47f8f36a0ea92e2140bed0105':
audioflinger: reduce sleep time to avoid underrun
|
| 7cafbb32999049873d4746ba83bd20c88abe6ce6 |
23-Nov-2011 |
Eric Laurent <elaurent@google.com> |
audioflinger: reduce sleep time to avoid underrun
Progressively reduce the sleep time applied in MixerThread::threadLoop()
in case of consecutive application underruns to avoid starving the audio HAL.
As the default sleep time is longer than the duration of an audio buffer
we ended up writing less data than needed by the audio HAL if
the condition persisted.
Issue 5553055.
Change-Id: I2b23ee79c032efa945025db228beaecd1e07a2e5
udioFlinger.cpp
|
| c23e2f2464eb3748599d47af7d8986b856f3c179 |
17-Nov-2011 |
Glenn Kasten <gkasten@google.com> |
Bug 5522189 re-enable ARM opts but disable inline
PIE did not work together with inlining. This change combines (almost
all of) the performance benefits of ARM optimizations together with PIE.
Change-Id: I4594d33ae5a0a7bac327ae08e30fb35343a06256
udioResampler.cpp
|
| c4795ecad4e5a0b3ec54862a40c82ef1ba53cd59 |
14-Nov-2011 |
Eric Laurent <elaurent@google.com> |
am 25924f8f: am 030bb998: Merge "audioflinger: fix noise when skipping to next song" into ics-mr1
* commit '25924f8f6c0a4ca4a2eb257b72d9625f69d2525e':
audioflinger: fix noise when skipping to next song
|
| 544fe9b6e9325701df4ab8c1d29774fc13c4cf6c |
12-Nov-2011 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix noise when skipping to next song
When audio effects are enabled, a noise can be heard at the
beginning of the new song when skipping to next song in music app.
This is because some effects (especially virtualizer) have a tail.
This tail was not played when previous song was stopped because effects were
not processed when no tracks were present on a given session. This is to
reduce CPU load when effects are enabled but no audio is playing.
The tail was then rendered when the new song was started.
Added a delay before stopping effect process after all tracks have been removed from a session.
Issue 5584880.
Change-Id: I815e0f7441f9302e8dfe413dc269a94e4cc6fd95
udioFlinger.cpp
udioFlinger.h
|
| 6977ca7d5ffdbc1610a95c74653b1fbe6a665f32 |
10-Nov-2011 |
Eric Laurent <elaurent@google.com> |
am db7d79e6: am 2b7f91b9: Merge "Fix regression for SoundPool playback" into ics-mr1
* commit 'db7d79e6f1e1860a9bfe4756a03c753435fd0ddf':
Fix regression for SoundPool playback
|
| a47b69c6f7c6fe0044ebcb2d0790ce3548de56fd |
09-Nov-2011 |
Eric Laurent <elaurent@google.com> |
Fix regression for SoundPool playback
Commit 19ddf0eb introduced a problem with applications (like SoundPool)
relying on an underrun condition to detect end of playback instead of
stopping the track when all data is written.
AudioFlinger would keep waiting for new data in case of partial buffer
filling and never reach the underrun condition.
Added a mechanism to wait no more than once if not enough frames are present
in the track buffer.
Issue 5585490.
Change-Id: I131e605ff6070831a01ddf734e68459e3bf2354b
udioFlinger.cpp
|
| 3b86c964df855a9740c446e984309b719c3ec37c |
08-Nov-2011 |
Eric Laurent <elaurent@google.com> |
am f3a892ab: Merge "AudioFlinger: mix track only when really ready" into ics-mr1
* commit 'f3a892ab9347ce733b81ccb4913a91c586f8f367':
AudioFlinger: mix track only when really ready
|
| 3dbe3201479828e84abe02e1fdd0a5d414c0ddb8 |
03-Nov-2011 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: mix track only when really ready
The addition of low power audio playback mode made that audio buffer consumption
by audio HAL can now happen in bursts. This makes that requesting audio data
from an AudioTrack for mixing can happen at much shorter intervals than before.
This revealed an existing problem where AudioFlinger would consider a track ready
for mixing although not enough frames were ready to completely fill one output buffer,
thus creating short periods of silence.
The fix consists in waiting for enough frames to be ready in AudioTrack buffer before
declaring a track ready for mixing. This minimum is not applied when the track is stopped
to allow the buffer to be emptied completely.
Change-Id: I6d04f9b65db5af85b0b53f0a5674be7ec02f9e9f
udioFlinger.cpp
|
| 305f4b6dabe500b97062e72049a296d0699243d7 |
28-Oct-2011 |
Glenn Kasten <gkasten@google.com> |
Bug 5522189 temporary workaround
Change-Id: I21d91d7a24df7bb6e7fc3d0fbc4786d55391fc0e
udioResampler.cpp
|
| 3856b090cd04ba5dd4a59a12430ed724d5995909 |
20-Oct-2011 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGV(_IF) to (IF_)ALOGV(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/#/c/143865
Bug: 5449033
Change-Id: I0122812ed6ff6f5b59fe4a43ab8bff0577adde0a
udioFlinger.cpp
udioMixer.cpp
udioPolicyService.cpp
udioResampler.cpp
|
| a85a74a8219c03f2b1d1ef98f3f02e55f89f89a3 |
19-Oct-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 381905: BassBoostTest CTS tests fail...
When AudioEffectTest is executed, an Equalizer is created
and enabled on a MediaPlayer session. Effects on the output
mix are therefore suspended.
Then the MediaPlayer is released with the effect still enabled.
In this case, Audioflinger::purgeStaleEffects_l() fails to restore
the suspended effects when the effect attached to the released audio session
is removed.
When subsequent tests are executed on output mix effects, these effects cannot be
enabled as they are still suspended.
Fixed purgeStaleEffects_l() to restore suspended effects if the effect removed is enabled.
Also fixed EffectHandle::disconnect() to only restore suspended effects if the disconnected
handle actually has control over the effect.
Change-Id: I67232e7c34680b0cc01abfd57d5d510a524e5d4f
udioFlinger.cpp
udioFlinger.h
|
| 5c4e818c39ac2d2739675fe907904a874f7623c5 |
19-Oct-2011 |
Eric Laurent <elaurent@google.com> |
Limit AudioFlinger log.
AudioFlinger logs a warning when a write to the audio HAL
takes too long to return. The threshold for this warning is
a rule of thumb based on the assumption that the audio HAL will consume
buffers at a regular pace.
The introduction of low power audio mode with larger buffers and writes
occuring in bursts makes that this threshold is often exceeded resulting
in excessive and misleading warnings.
The threshold is raised to remove unwanted warnings but we should reconsider
the usefulness of this warning altogether.
Change-Id: I5ef6898ea28d879cede3e47da542a64092a3cca4
udioFlinger.cpp
|
| ec35a1416472865dbebc22b10199ad718ed2cc95 |
06-Oct-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 5381089: problem with A2DP music volume
This problem only occurs when audio effects are present and
the music volume is applied by one effect engine.
When connecting or disconnecting A2DP, audio effects are moved from
one mixer thread to another. When removed from the source thread,
the effect is stopped but it is not restarted when added to the
destination thread.
This regression was introduced by commit 21b5c47e.
Change-Id: I4cc578d8d760ec65b185032b6fda98c739d331bc
udioFlinger.cpp
udioFlinger.h
|
| 9d18ec574f5d847a86a21594ac39394ef5b108fc |
27-Sep-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 5373658: memory leak in AudioFlinger.
Unlink PowerManagerService binder interface death recipient
in ThreadBase destructor.
Change-Id: Iab06ae9a8a6737bb002b6416a157b0fb50c11ad5
udioFlinger.cpp
|
| 6bffdb8b598a2399e57f6ca48660fb7bdb2490b5 |
23-Sep-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 5355047: Automated effect tests fail.
Fixed several regressions in automated audio effect tests due
to changes in effect framework and visualizer FFT output range.
- Do not suspend Volume effect on session 0 when effects are
enabled on specific sessions.
- Adapt energy detection thresholds to new visualizer FFT range.
- Leave more time for BassBoost and Virtualizer effects to ramp up
before measuring the effect.
- Removed second insert reverb left by mistake on the player session
in preset reverb test.
Change-Id: I7a1ad1372d783fa7900eb9dd1d3b47f54d8d766f
udioFlinger.cpp
|
| eb8b914ad9d4331e1cdf4346731770ce69fd0e77 |
16-Sep-2011 |
Nick Kralevich <nnk@google.com> |
AudioResampler: use relative instead of absolute labels
This change is needed to allow Android to compile with -fPIE
Bug: 5328392
Change-Id: I84d947975776800a7b79c6ac75a881af461a631c
udioResampler.cpp
|
| 60cd0a0d488e604d27fc7dbb02b00348693dfde2 |
13-Sep-2011 |
Eric Laurent <elaurent@google.com> |
Issue 4345021: Audio routed to multiple devices...
There is a possiblility that the condition on which RecordThread::checkForNewParameters_l()
waits after updating the command completion status is never signalled.
This happens if the thread executing ThreadBase::setParameters() has timed out waiting
for the status (for instance if the audio HAL takes too long to execute the setParameters()
command. Then the RecordThread is stuck forever.
The fix consists in waiting for the condition with a timeout in RecordThread::checkForNewParameters_l().
Change-Id: I7fc671bc2fc43ba4acb65a2beb33ee05742f091e
udioFlinger.cpp
|
| b76e90de3c64626fe07a68469d0a59a31c8efb6b |
30-Aug-2011 |
Eric Laurent <elaurent@google.com> |
Merge "226483: A2DP connected, but music out to speaker"
|
| 9f6530f53ae9eda43f4e7c1cb30d2379db00aa00 |
30-Aug-2011 |
Eric Laurent <elaurent@google.com> |
226483: A2DP connected, but music out to speaker
When the A2DP headset is connected, there is a possible
race condition when the audio tracks are moved from
the mixer thread attached to the speaker output to the thread
attached to A2DP output.
As the request to clear the stream type to output mapping cache in
the client process is asynchronous, it is possible that the flag
indicating to the client audio track to re-create the IAudioTrack
on the new thread is processed before the cache is invalidated.
In this case, the track will be attached to the old thread and
music will continue playing over the device speaker instead of being
redirected to A2DP headset.
Change-Id: Ib2ce1eb5320eaff83287b93779061bf4e7a330df
udioFlinger.cpp
udioFlinger.h
|
| bee5337da7659b3b7128622ba1f42618b11df5be |
29-Aug-2011 |
Eric Laurent <elaurent@google.com> |
Audioflinger: reverse logic of BT NREC indication
The interpretation of BT NREC by AudioFlinger to enable
or disable AEC and NS was wrong: NREC to ON (default) means
the phone (Audio Gateway) must enable local AEC and NS.
Change-Id: I88a264e7fc9831c43bbace4f6b585baec73f2006
udioFlinger.cpp
udioFlinger.h
|
| a1884f9e9ec3836683efd7eb333ee442e8bc9d56 |
23-Aug-2011 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: add check for audio HAL init failure
Do not call audio HAL functions on the primary HW interface
if it could not be initialized properly.
Change-Id: If54059c8fd188d6c1686f9e0439994fe9411478a
udioFlinger.cpp
|
| a5cc7cce9b8aee73b08f6532710e186c02fdd1c0 |
03-Aug-2011 |
Eric Laurent <elaurent@google.com> |
Issue 5081351: isWiredHeadsetOn() permission.
AudioManager.isWiredHeadsetOn() should not require permission MODIFY_AUDIO_SETTINGS.
Remove permission checks on all getters in audio policy manager as permission enforcement
is really usefull for setters.
Also deprecate AudioManager.isWiredHeadsetOn() which name and implementation are deceptive.
Change-Id: I38f8df7c26c0d417bf0e2b74e4c11c2d143f2ecd
udioPolicyService.cpp
|
| 0270b188aa3929cc512ec6869caba1d6b60cc08c |
12-Aug-2011 |
Marco Nelissen <marcone@google.com> |
Don't remove effects on session 0
b/5159902
Change-Id: Id6f29fb1c687069f7480dd81d4745a558f202226
udioFlinger.cpp
|
| db7c079f284f6e91266f6653ae0ec198b1c5006e |
10-Aug-2011 |
Eric Laurent <elaurent@google.com> |
Audio effects: track CPU and memory use separately
Before this change, CPU and memory usage for an audio effect were
registered and checked against the limit by audio policy manager
upon effect instantiation. Even if an effect was not enabled
it would prevent another effect to be created if the CPU load budget
was exceeded, which was too restrictive.
This change adds a method to register/unregister CPU load only when
an effect is enabled or disabled.
It also adds a mechanism to place all effects on the global output mix
in suspend state (disabled) when an effect is enabled on a specific session.
This will allow applications using session effects to have the priority
over others using global effects.
Also fixes some issues with suspend/restore mechanism:
- avoid taking actions when an effect is disconnected and was not enabled.
- do not remove a session from the suspended sessions list when corresponding
effect chain is destroyed.
Change-Id: I5225278aba1ae13d0d0997bfe26a0c9fb46b17d3
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
udioPolicyService.h
|
| 3a34befc6fb04a4945a849e8bda8b84e4bf973fe |
02-Aug-2011 |
Marco Nelissen <marcone@google.com> |
Keep effects sessions active when the caller dies.
Don't remove effects until the session they are in goes away or all
AudioEffects have been explicitly released. This allows the control
panel process to die without stopping the effects.
Change-Id: I4496e5df080230ca1af149dec95c1309ab8ea888
udioFlinger.cpp
udioFlinger.h
|
| c3e6572e0ff535932b1f6ffb7bcf5acd891675fb |
08-Aug-2011 |
Eric Laurent <elaurent@google.com> |
Merge "AudioFlinger: protect input/output stream access"
|
| b8ba0a979067a4efb0b3819bf17770793e41c15e |
08-Aug-2011 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: protect input/output stream access
Some methods would not check that the output orinput stream of a thread
was still valid before calling functions on its interface.
This could cause a crash if those methods where called while the output or
input was being closed by another thread.
Make sure that the output or input stream pointer is cleared before closing the
stream.
Always check that the output or input pointer is not null before calling
functions at the stream interface.
Generalize the use of initCheck() method to verify that the output or input
stream is not null.
Change-Id: I9d9ca6b744d011bcf3a7bbacb4a581ac1477bfa5
udioFlinger.cpp
udioFlinger.h
|
| 59bd0da8373af0e5159b799495fda51e03120ea4 |
01-Aug-2011 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: disable AEC and NS with BT headsets
Disable AEC and NS when the Bluetooth SCO headset in use indicates it
implements those pre processings.
Change-Id: I93f3d10b0a27243d5dbff7182639576fc0c6d862
udioFlinger.cpp
udioFlinger.h
|
| 59255e4fc7d8ff52874b85b1988dc0785140cf81 |
28-Jul-2011 |
Eric Laurent <elaurent@google.com> |
Audio Effect Framework: add effect suspend/restore
Add the possibility for the effect framework to suspend
(temporarily disable process) and restore audio effects.
This feature will be usefull to disable pre processing under certain
conditions and better control coexistence of audio effects
on output mix and specific sources.
Change-Id: I79b195982cc48748d5708308fb1647b9c3c34cc6
udioFlinger.cpp
udioFlinger.h
|
| 6dbe883644940badc684957cfc381bfd115f205e |
28-Jul-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 5090721: audio record broken
Commit 6dbdc40 introduced a deadlock when exiting the
AudioFlinger RecordThread.
Change-Id: I1f63e54c5aeff05da4e4d028b53f734c62c78677
udioFlinger.cpp
|
| a7280a59259018d997896c043fd2db95f631f12e |
27-Jul-2011 |
Eric Laurent <elaurent@google.com> |
Merge "AudioFlinger: fix crash when deleting pre process."
|
| ec437d8d3db79459d7b19e1734e6fe309bd621e8 |
27-Jul-2011 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: fix crash when deleting pre process.
If a pre processing effect is detroyed while enabled and capture is active,
there was a possibility that the effect engine is released by the framework
while still processed by the audio HAL.
The fix consists in not releasing the engine in EffectModule::removeHandle()
but just flag the effect as being detroyed to avoid further calls to functions
on the engine effect interface.
The effect interface is then removed from the audio HAL safely in
EffectChain::removeEffect_l() while holding the EffectChain mutex.
Change-Id: I71fab30d9145062af8644f545a1f1d4d3e7e7f02
udioFlinger.cpp
udioFlinger.h
|
| feb0db689c17dced50afaee54c659f1676e2d505 |
22-Jul-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 4604090: notification sound interrupted.
The problem is that the audio HAL fails to acquire the wake lock when playing the notification.
This is because of a change that removed the mediaserver process form the system group for honeycomb.
The fix consists in requesting the wake lock from PowerManagerService when AudioFlinger mixer
wakes up.
A consequence of this change is that audio HALs or pcm drivers do not have to hold wake locks
anymore as in the past.
Change-Id: I4fb3cc84816c9c408ab7fec75886baf801e1ecb5
ndroid.mk
udioFlinger.cpp
udioFlinger.h
|
| 57dae99c9fcd6becd2b5ed0c53f277ba4d7dbcfc |
24-Jul-2011 |
Eric Laurent <elaurent@google.com> |
Added APIs for audio preprocessing
Added APIs to control pre processes applied on captured audio.
Those APIs are still hidden until reviewed by API council.
Three types of standard pre processes are supported:
- Automatic Gain Control (AGC) by AutomaticGainControl class
- Acoustic Echo Cancellation (AEC) by AcousticEchoCanceler class
- Noise Suppression (NS) by NoiseSuppressor class
A method is added to AudioEffect class to query audio pre processings
applied by default by the platform on a given AudioRecord session ID.
Change-Id: I0b9fceeb8c704dd06319c3b52b85c96fe871d51d
udioPolicyService.cpp
udioPolicyService.h
|
| 1d2bff0e588afe183a1baaae731519b4e957bbdb |
25-Jul-2011 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: add dump of audio pre processing.
Dump of media.audio_flinger service was only listing effects on output threads.
Moved the dump of effect chains from PlaybackThread to ThreadBase class so that
pre processings on RecordThread are also listed.
Change-Id: If8bc74023c12b9c2371f1b300743b156ceca7b87
udioFlinger.cpp
udioFlinger.h
|
| edc15ad8fcde12dc4f642d80d077239b1532eeca |
22-Jul-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 4988574: 8 bit PCM audio playback broken.
Fixed regression in audio track control block frame size
calculation introduced by commit c310dcb.
Change-Id: Ia731b946ae4e43316b98d80229e3b08a696e47d6
udioFlinger.cpp
|
| 7c7f10bd4fda9a084e5e7f0eb3a040dfcbf01745 |
18-Jun-2011 |
Eric Laurent <elaurent@google.com> |
Audio framework: support for audio pre processing
Audio effect framework is extended to suport effects on
output and input audio path.
AudioFlinger: Support for audio effects and effect chains is
moved from PlaybackThread class to ThreadBase class so that
RecordThread can manage effects.
Effects of type pre processing are allowed on record thread
only. When a pre processing is enabled, the effect interface handle is
passed down to the input stream so that the audio HAL can call the
process function. The record thread loop calls the effect chain process
function that will only manage the effect state and commands and skip the
process function.
AudioRecord: The audio session is allocated before calling getInput() into
audio policy serice so that the session is known before the input theead is
created and pre processings can be created on the correct session.
AudioPolicyService: default pre processing for a given input source are
loaded from audio_effects.conf file.
When an input is created, corresponding effects are created and enabled.
Change-Id: Id17119e0979b4dcf189b5c7957fec30dc3478790
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
udioPolicyService.h
|
| e0b5bb23f0a26d248275d203885b820659da7320 |
16-Jul-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Log CPU usage"
|
| 4d8d0c30abfa4b8d75866d42094cc797e05068fa |
09-Jul-2011 |
Glenn Kasten <gkasten@google.com> |
Log CPU usage
Change-Id: Ie447e59be139153e526b7ad467c46c659d26816f
ndroid.mk
udioFlinger.cpp
|
| 81eedd1746be590d5d506dfb9ff23619f9d9898f |
13-Jul-2011 |
Mike Lockwood <lockwood@android.com> |
Remove some #ifdef HAVE_ANDROID_OS that were needed for the simulator build
Change-Id: I13d9f251f86c05ae5405f37adbf6b8e9660935ba
Signed-off-by: Mike Lockwood <lockwood@android.com>
udioPolicyService.cpp
|
| 5e0067b486c3862316aa1f293cf9690c0cf54bda |
12-Jul-2011 |
Jeff Brown <jeffbrown@google.com> |
Remove the simulator target from all makefiles.
Bug: 5010576
Change-Id: I04d722f258951a3078fe07899f5bbe8aac02a8e8
ndroid.mk
|
| 6c21234e3d254fc58227cd2063500a0dcd173689 |
08-Jul-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix build warning; LOG_TAG was misplaced"
|
| 5159c7ec97b575c0486159f0543d2da90c3c3b68 |
08-Jul-2011 |
Glenn Kasten <gkasten@google.com> |
Fix build warning; LOG_TAG was misplaced
Change-Id: I4fb0590f626445188f9053367707878b2fa0fa30
udioResamplerCubic.cpp
|
| 6d8b694d999e9be7d5dcc336535832a80fb6f61f |
24-Jun-2011 |
Eric Laurent <elaurent@google.com> |
Moved and renamed effect API header files
Moved specific effect header files to
system/media/audio_effects/include/audio_effects
and renamed to lower case (effect_xxx.h).
Change-Id: Icfc2264bfd013cab0395d7e310ada636b9fe3621
ndroid.mk
udioFlinger.cpp
|
| 671a636931295d9c33ffca74551a804479d01241 |
17-Jun-2011 |
Eric Laurent <elaurent@google.com> |
Added audio_bytes_per_sample() helper function
Change-Id: Ibfcd75c4c241a53d5f052c25ada091904991048a
udioFlinger.cpp
|
| 7394a4f358fa9908a9f0a7c954b65c399f4268e6 |
14-Jun-2011 |
Dima Zavin <dima@android.com> |
audio: update for audio/audio_policy header names/locations
Change-Id: I36c49352eee57559403cd1597f56a8485a360289
Signed-off-by: Dima Zavin <dima@android.com>
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
udioPolicyService.h
|
| 0512ab559d4670c2204078470d7ef5d376811c57 |
05-May-2011 |
Glenn Kasten <gkasten@google.com> |
Remove dead code related to gettid
The gettid system call is always available now.
Change-Id: Ib78b41781eda182dc8605daf456bbea7ff7c2dc0
udioFlinger.cpp
udioPolicyService.cpp
|
| 0d255b2d9061ba31f13ada3fc0f7e51916407176 |
25-May-2011 |
Jean-Michel Trivi <jmtrivi@google.com> |
Use channel mask instead of channel count for track creation
Record and playback objects (resp AudioRecord and AudioTrack)
are created using a channel mask, but this information is lost
in the mixer because only the channel count is known to
AudioFlinger. A channel count can always be derived from a
channel mask.
The change consists in:
- disambiguiting variable names for channel masks and counts
- passing the mask information from the client to AudioFlinger
and the mixer.
- when using the DIRECT ouput, only verifying the format of
the track is compatible with the output's for PCM.
Change-Id: I50d87bfb7d7afcabdf5f12d4ab75ef3a54132c0e
udioFlinger.cpp
udioFlinger.h
udioMixer.cpp
udioMixer.h
|
| 65580f9adf6c4d98449ad0716488f9fe3869aa5a |
28-May-2011 |
Eric Laurent <elaurent@google.com> |
Removed interface to load audio effects libraries
Removed unused functions allowing dynamic loading of audio effects libraries
from effects factory API.
Change-Id: I06cc5a51dc10aca87c7a8687bbb874babd711eca
udioFlinger.cpp
udioFlinger.h
|
| e1315cf0b63b4c14a77046519e6b01f6f60d74b0 |
18-May-2011 |
Eric Laurent <elaurent@google.com> |
New effect library API
Moved and renamed media/EffectApi.h to hardware/audio_effect.h
Modified the effect library API to expose a library info structure
containing an interface functions table.
Also removed enums for audio channels, audio format and devices
from effect API and use values from system/audio.h instead.
Modified effects factory to support new library interface format and
load libraries and efffects listed in audio_effects.conf file.
The file audio_effects.conf is first loaded from /vendor/etc and
then from /system/etc/audio_effects.conf if not found.
Modified existing effect libraries to implement the new library interface.
Change-Id: Ie52351e071b6d352fa2fbc06c3846686f8c45df9
udioFlinger.cpp
udioFlinger.h
|
| 162f7d15ac5c8c23d1c3de171239f3a4e6e06b2a |
23-May-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "HAVE_ANDROID_OS AUDIOFLINGER_SECURITY_ENABLED dead"
|
| 6708b9a3fb654f5623ba5a696288fdba310a5e1a |
13-May-2011 |
Eric Laurent <elaurent@google.com> |
Merge "Fix audio effect framework issues"
|
| b469b9490b3cd9e0f0466d9b9ab228f6c793b82e |
09-May-2011 |
Eric Laurent <elaurent@google.com> |
Fix audio effect framework issues
Fix two issues in audio effect framework reported by partners.
1 - Fixed duplicated audio buffer sent to effect process function when
pausing a track.
Modified Effectchain::process_l() function to clear the effect chain
input buffer before calling the effect process functions when no track
is active on the session. Previous code was clearing the buffer after
calling the process functions and when transitioning from active
to inactive, the last processed buffer was passed again once to effect
process function before being cleared.
2 - Fixed potential mutex cross deadlock when disconnecting an effect
while playback is active. This is because EffectChain::process_l()
was calling PlaybackThread::hasAudioSession() thus creating an inversion
in the mutex lock order (EffectChain mutex locked before ThreadBase mutex).
The fix consists in removing the call to hasAudioSession() from process_l()
and requires each effect chain to keep count of the number of audio tracks
attached to it (previously only the active tracks were accounted for).
Change-Id: Iee4246694ea8c7a66c012120c629d72dd38f9c35
udioFlinger.cpp
udioFlinger.h
|
| 64760240f931714858a59c1579f07264d7182ba2 |
11-May-2011 |
Dima Zavin <dima@android.com> |
update for new audio.h header location
Change-Id: Ic4c62c4037800802427eb7d3c7f5eb8b25d18876
Signed-off-by: Dima Zavin <dima@android.com>
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
udioPolicyService.h
|
| 249c6a61f21bc90e25e4b77f18c98af1ac363e69 |
05-May-2011 |
Glenn Kasten <gkasten@google.com> |
HAVE_ANDROID_OS AUDIOFLINGER_SECURITY_ENABLED dead
Remove dead code from the days when Android ran in emulator.
Change-Id: Ibadbbde0538239ad9b2811a3a2e8f8a6d3b6389c
udioFlinger.cpp
|
| 5a61d2f277af3098fc10b2881babca16391362da |
20-Apr-2011 |
Dima Zavin <dima@android.com> |
audioflinger: don't do work in constructor, instead do it in onFirstRef
Change-Id: I22d9e01821816c3beb52b014330386c7fd2f0411
Signed-off-by: Dima Zavin <dima@android.com>
udioFlinger.cpp
udioFlinger.h
|
| 799a70e7028a4d714436c3a744a775acfbd31aae |
19-Apr-2011 |
Dima Zavin <dima@android.com> |
audioflinger: enumerate all the possible audio interfaces
Keep track of the primary interface that handles the master volume,
etc.
Change-Id: Ib0701fccff8d8783a99035a241ab7c8ec75c00ac
Signed-off-by: Dima Zavin <dima@android.com>
udioFlinger.cpp
udioFlinger.h
|
| fce7a473248381cc83a01855f92581077d3c9ee2 |
20-Apr-2011 |
Dima Zavin <dima@android.com> |
audio/media: convert to using the audio HAL and new audio defs
Change-Id: Ibc637918637329e4f2b62f4ac7781102fbc269f5
Signed-off-by: Dima Zavin <dima@android.com>
ndroid.mk
udioFlinger.cpp
udioFlinger.h
udioPolicyService.cpp
udioPolicyService.h
|
| 6b80e0be94d3f92ec4aa2b7cace816780d3f338d |
20-Apr-2011 |
Dima Zavin <dima@android.com> |
audioflinger: move legacy audio hw/policy out to libhardware_legacy
Change-Id: I4adcec73d3c08bcbe15bb19e1ba2ff18b195af45
Signed-off-by: Dima Zavin <dima@android.com>
2dpAudioInterface.cpp
2dpAudioInterface.h
ndroid.mk
udioDumpInterface.cpp
udioDumpInterface.h
udioHardwareGeneric.cpp
udioHardwareGeneric.h
udioHardwareInterface.cpp
udioHardwareStub.cpp
udioHardwareStub.h
udioPolicyManagerBase.cpp
|
| dab0bea38ab2c2d6b52907b7f5684834c4979251 |
26-Apr-2011 |
Eric Laurent <elaurent@google.com> |
am 48dca4de: am 969357e3: am 223995dc: Fix issue 4335692: HDMI media volume
* commit '48dca4de70890d324b5830a58bb9fa273164151a':
Fix issue 4335692: HDMI media volume
|
| 5bfdfd1e7b4514c6f6143b6b97ef37013f6e53a9 |
26-Apr-2011 |
Eric Laurent <elaurent@google.com> |
am 969357e3: am 223995dc: Fix issue 4335692: HDMI media volume
* commit '969357e31c93e168e7a4cb3b41c33a6ec26f4a26':
Fix issue 4335692: HDMI media volume
|
| 023eeac54c428d78b755fa3c856f20d17abb8a40 |
25-Apr-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 4335692: HDMI media volume
Forced music stream volume to max when not muted and output device is HDMI.
Change-Id: Ibd287cea8ae1d3f36fea6651a113bd5cf2dbad13
udioPolicyManagerBase.cpp
|
| 5820e52f9bf3c40839d33457975b6c9629ab1c79 |
12-Apr-2011 |
Brad Fitzpatrick <bradfitz@android.com> |
am 07c25b65: am 4430670f: am ca7ad44a: Merge "audioflinger: Enable ARMv5TE optimized resampler"
* commit '07c25b651216daac884bd503c02b788418165b50':
audioflinger: Enable ARMv5TE optimized resampler
|
| 4d0243980123378e9dd69022bf4ddca729d84126 |
12-Apr-2011 |
Brad Fitzpatrick <bradfitz@android.com> |
am 4430670f: am ca7ad44a: Merge "audioflinger: Enable ARMv5TE optimized resampler"
* commit '4430670f48f75661293371dab8db55865f5f56e0':
audioflinger: Enable ARMv5TE optimized resampler
|
| 0c0a1c0c37dbd2646a732da706d6777283c83e44 |
06-Apr-2011 |
Jim Huang <jim.huang@linaro.org> |
audioflinger: Enable ARMv5TE optimized resampler
Previously, the optimized asm option is only enabled when
__ARM_ARCH_5E__ is defined, which is assigned in armv5te.mk
rather than armv7-a series targets. This patch checks the ARM CPU
feature about half-word multiply instructions to enable ARMv5TE
resampler optimization routines properly.
Change-Id: I4c5a5d8c932416f23bedb0b389db958349f21ea4
udioResampler.cpp
|
| c30268b9d118309a0514bcf280a03ee69f81403f |
06-Apr-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Miscellaneous code cleanup in audio framework"
|
| 4bcae82f9b07d1a39956c45a6f5bec0b696c4dd1 |
04-Apr-2011 |
Glenn Kasten <gkasten@google.com> |
Miscellaneous code cleanup in audio framework
Changes:
- Move declaration of kClassPathName to top of file so it can be used
in more than one place, instead of "android/media/AudioSystem".
- Make private methods static.
- Add comment to stream_type, audio_mode, force_use types that they must match
values in AudioSystem.java.
- Add comment about unused types mp3_sub_format and vorbis_sub_format.
- Fix typos.
- Use @ in javadoc comments.
- Delete dead APIs setMode, getMode, setRouting, getRouting in AudioSystem.java
(they are all hidden, deprecated, and unused by rest of framework)
- Delete unused private log method.
- Fix pathname for android_media_AudioSystem.cpp.
- Improve code formatting for space after == and !=.
- Add logging of delta for changing audio policy manager ref count.
Change-Id: I18037c7beb8ab76d1fda08c11e589f6e591d36e1
udioFlinger.cpp
udioPolicyManagerBase.cpp
|
| 38ccae2c0324daa305f3fe77d25fdf5edec0b0e1 |
29-Mar-2011 |
Eric Laurent <elaurent@google.com> |
New fix for issue 4111672: control block flags
The first fix (commit 913af0b4) is problematic because it makes threads
in mediaserver process block on the cblk mutex. This is not permitted
as it can cause audio to skip or worse have a malicious application
prevent all audio playback by keeping the mutex locked.
The fix consists in using atomic operations when modifying the control
block flags.
Also fixed audio_track_cblk_t::framesReady() so that it doesn't block
when called from AudioFlinger (only applies when a loop is active).
Change-Id: Ibf0abb562ced3e9f64118afdd5036854bb959428
udioFlinger.cpp
|
| 33797ea64d067dfeaacbfd7ebe7f3383b73961b5 |
17-Mar-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 4111672: AudioTrack control block flags
Make sure that all read/modify/write operations on the AudioTrack
and AudioRecord control block flags field are protected by the
control block's mutex.
Also fix potential infinite loop in AudioTrack::write() if the
written size is not a multiple of frame size.
Change-Id: Ib3d557eb45dcc3abeb32c9aa56058e2873afee27
udioFlinger.cpp
|
| ce80563f6fb49b0c40281f9a527e288799b4a577 |
15-Mar-2011 |
Eric Laurent <elaurent@google.com> |
Improvement for issue 3489986: BT SCO volume
This change makes sure that the VOICE_CALL stream volume tracks
the BLUETOOTH_SCO stream volume when SCO audio is enabled.
The down link audio volume now reflects what is being displayed
when pressing volume hard keys on the device while in a video chat
with a BT SCO headset.
Volume settings on the headset and the device are still independent as
we do not support handsfree profile yet.
Change-Id: Ie0d2714730ea359b9318b9cbe6f0b2557ef0f976
udioPolicyManagerBase.cpp
|
| 4f46fed004b76d462ef859aa94a3c056b0e79b60 |
03-Mar-2011 |
Gloria Wang <gwang@google.com> |
Merge "- To track the usage of all audio output devices - To track the currently used audio device - The devices are separated as speaker and other audio devices - Provide the collected data to battery application through pullBatteryData()"
|
| 9ee159b79022b2e1a050acb3890ce948e99e9ccb |
24-Feb-2011 |
Gloria Wang <gwang@google.com> |
- To track the usage of all audio output devices
- To track the currently used audio device
- The devices are separated as speaker and other audio devices
- Provide the collected data to battery application through pullBatteryData()
Change-Id: I374c755266b5ac6b1c6c630400f4daf901ea8acc
udioFlinger.cpp
|
| 1c4967155109ab083a4149347596f7ebd3662a67 |
03-Mar-2011 |
Eric Laurent <elaurent@google.com> |
Merge "Fix issue 3388775."
|
| c0bff9b96152179650dc9508799a38e35ef2a321 |
01-Mar-2011 |
Eric Laurent <elaurent@google.com> |
Merge "Fix issue 3479042."
|
| 532492cd60a3ee2613cd3d664c9f4581d9929bc5 |
01-Mar-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 3388775.
Do not select A2DP output for media strategy when it is suspended because
BT SCO is active. Media audio will be routed to speakers or SCO HS
(depending on phone state and activity on stream VOICE_CALL) which is less
confusing than not hearing anything while music progress bar is moving.
Change-Id: Iff8cc1ea9bf9bde0b33035c4d91398db0934b836
udioPolicyManagerBase.cpp
|
| 243f5f91755c01614a8cafe90b0806396e22d553 |
01-Mar-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 3479042.
The problem is that when an AudioRecord using the resampler is restarted,
the resampler state is not reset (as there is no reset function in the resampler).
The consequence is that the first time the record thread loop runs, it calls the resampler
which consumes the remaining data in the input buffer and when this buffer is released
the input index is incremented over the limit.
The fix consists in implementing a reset function in the resampler.
A similar problem was also present for playback but unoticed because the track buffer is always
drained by the mixer when a track stops. The only problem for playback was that the initial
phase fraction was wrong when restarting a track after stop (it was correct after a pause).
Change-Id: Ifc2585d685f4402d29f4afc63f6efd1d69265de3
udioFlinger.cpp
udioMixer.cpp
udioMixer.h
udioResampler.cpp
udioResampler.h
|
| 043b22dbe3b98a8fdd6b0ea77df4089ff44b6539 |
28-Feb-2011 |
Jean-Michel Trivi <jmtrivi@google.com> |
Fix bug 3376700 Volume too loud at lowest setting
Change volume attenuation curve to provide more attenuation at
low volume settings, and finer steps at high volume.
See bug entry for link to doc with curve values.
Change-Id: I750548b2161a4c550ef982ba793156e4518119e8
udioPolicyManagerBase.cpp
|
| 586b21780680c4d19723b5a9484e55ec47464c5f |
11-Feb-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 3400751.
Add a delay before restoring output path when a notification ends so that
short sounds can be heard on proper device before the path is actualy switched.
Change-Id: I1d2dd8e7e28e15fbcab344256f88499b26297372
udioPolicyManagerBase.cpp
|
| 727a6f859841a1d421b680f56d4d4ca5ff8c6e4b |
11-Feb-2011 |
Eric Laurent <elaurent@google.com> |
Merge "Fix issue 3425342."
|
| 12932bfdc7163e654d5ef25f1cb1802ded2704f1 |
11-Feb-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 3425342.
Change the device selection order as follows to enable easier use of
A2DP while the device is docked:
1 - wired Headset
2 - A2DP Headset
3 - SPDIF/HDMI
4 - Dock
Also do not limit notifications volume when on dock.
Change-Id: I55ea6bea9f2d9ff284b54023e541b2788d0f1eb8
udioPolicyManagerBase.cpp
|
| 6b2718c67aa7b1a8e3b0f25a73a0d5f72c59ffc3 |
04-Feb-2011 |
Glenn Kasten <gkasten@google.com> |
Bug 3352047 Wrong message when adjusting volume
Add hidden AudioManager.getDevicesForStream and output device codes.
Change-Id: I4d1c1d3b6a077cd117720817d1f733dda557b947
udioPolicyManagerBase.cpp
udioPolicyService.cpp
udioPolicyService.h
|
| 8473bb556421ac08f4e03abd5d149852114c9238 |
07-Feb-2011 |
Brad Fitzpatrick <bradfitz@android.com> |
am be806fe8: am cc8f87e9: am f664d6f9: am b083d3b8: Merge "Initialize resampling buffer per track."
* commit 'be806fe8c1c7bb3ae70ae27dce41d672410af26a':
Initialize resampling buffer per track.
|
| 802da7bc462f6f9df6e681cc75c51dc0a056a4a1 |
04-Feb-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Bug 3366668 Use BinderService template"
|
| e08d819989d394ed171ba6f1e6d0fa14c29fc42d |
04-Feb-2011 |
Brad Fitzpatrick <bradfitz@android.com> |
am cc8f87e9: am f664d6f9: am b083d3b8: Merge "Initialize resampling buffer per track."
* commit 'cc8f87e9410dd4de9a2fda4738429e6c6087c789':
Initialize resampling buffer per track.
|
| e26308ad434c575b2985ac442d2990bb1bc19f5d |
04-Feb-2011 |
Brad Fitzpatrick <bradfitz@android.com> |
am b083d3b8: Merge "Initialize resampling buffer per track."
* commit 'b083d3b816378ef3b9dceb33b2c2e20510b2632b':
Initialize resampling buffer per track.
|
| 2151d7b8c2dd77c9887691db30396937be778141 |
04-Feb-2011 |
Yuuhi Yamaguchi <yuuhi.x.yamaguchi@sonyericsson.com> |
Initialize resampling buffer per track.
When resampling too short sound, AudioMixer uses previous
tracks buffer. So we re-initialize the temporary buffer per
loop to avoid it.
Change-Id: I55a59a3b14faa8445e09c450478fe79cef704760
udioMixer.cpp
|
| 117cd9286424888c1c5bf202ebf1e08ae1e6affe |
04-Feb-2011 |
Glenn Kasten <gkasten@google.com> |
Merge "Bug 3366885 Remove LVMX switch"
|
| 8782cb4dc783cdcac8498148ab1894b4780c18df |
04-Feb-2011 |
Eric Laurent <elaurent@google.com> |
am 6f1bd261: am 9c0a1003: Merge "Fix issue 3371080" into honeycomb
* commit '6f1bd261b7fd86ac7817ca061dfb55b95150b836':
Fix issue 3371080
|
| db130fbd3ccd37e247e49494a84f8a9841ecd593 |
04-Feb-2011 |
Glenn Kasten <gkasten@google.com> |
Bug 3366885 Remove LVMX switch
Change-Id: I0bf98c6f85f00b3296874571e1c049dcc4e2fcca
ndroid.mk
udioFlinger.cpp
udioFlinger.h
|
| d2dcb08808aed82218b258726fa66f8e095a3f9c |
04-Feb-2011 |
Glenn Kasten <gkasten@google.com> |
Bug 3366668 Use BinderService template
Change-Id: I93d7f3fc9dc9b6a365723d8a51a73a5aabdb4f93
udioPolicyService.cpp
udioPolicyService.h
|
| eda6c364c253ba97ee45a3adeb8c2b45db1f81db |
02-Feb-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 3371080
Modified default volume control logic in AudioService:
1 IN_CALL volume if in video/audio chat
2 NOTIFICATION if notification is playing or was playing less than 5s ago.
3 MUSIC
Modified silent mode:
- now also affect MUSIC stream type
- entering silent mode when VOL- hard key is pressed once while selected
stream volume is already at 0 (except for VOICE_CALL stream).
- exiting silent mode when pressing VOL+ hard key while in silent mode
Play sound FX (audible selections, keyboard clicks) at a fixed volume.
Modified audio framework:
- isStreamActive() method now implemented in AudioPolicyManagerBase (previously AudioFlinger)
- iStreamActive() now specifies a time window during which the stream is considered
active after it actually stopped.
Change-Id: I7e5a0724099450b9fc90825224180ac97322785f
udioFlinger.cpp
udioFlinger.h
udioPolicyManagerBase.cpp
udioPolicyService.cpp
udioPolicyService.h
|
| 2a30be33a39b07bf4f945d9506ffda0fd076e6c2 |
29-Jan-2011 |
Jean-Baptiste Queru <jbq@google.com> |
Merge 0ef57993 from gingerbread
Change-Id: If10fee1ae387a8130356dd62fe678495402d5edf
|
| 144f78906f1eefaf8cbde9298a1422fcea094447 |
29-Jan-2011 |
Jean-Baptiste Queru <jbq@google.com> |
am 4eeb1047: Merge 13212f83 from gingerbread-plus-aosp
* commit '4eeb10470ffafe8c508027f363ac66b58da5bf00':
Fix issue 2988031.
|
| 2ef89309d41daac3d69fa78b31cd09cb2c262ca1 |
29-Jan-2011 |
Jean-Baptiste Queru <jbq@google.com> |
Merge 13212f83 from gingerbread-plus-aosp
Change-Id: I9a8ee0c7e7896aea85e7a7c18ee82927091cb670
|
| 5092d8c066a1a98343bbc16e33a6753e577b5e7f |
27-Jan-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 2988031.
Limit SYSTEM stream volume when a headset is connected and music is playing.
Change-Id: Ieb44ae5bb53ffa9cd5fe8e317798eed279b78df8
udioPolicyManagerBase.cpp
|
| bfa2f13fd3f463dbceea4d3a18c3124e70df0a05 |
15-Nov-2010 |
Jean-Michel Trivi <jmtrivi@google.com> |
do not merge bug 3370834 Cherrypick from master
Cherripick from master CL 79833, 79417, 78864, 80332, 87500
Add new audio mode and recording source for audio communications
other than telelphony.
The audio mode MODE_IN_CALL signals the system the device a phone
call is currently underway. There was no way for audio video
chat or VoIP applications to signal a call is underway, but not
using the telephony resources. This change introduces a new mode
to address this. Changes in other parts of the system (java
and native) are required to take this new mode into account.
The generic AudioPolicyManager is updated to not use its phone
state variable directly, but to use two new convenience methods,
isInCall() and isStateInCall(int) instead.
Add a recording source used to designate a recording stream for
voice communications such as VoIP.
Update the platform-independent audio policy manager to pass the
nature of the audio recording source to the audio policy client
interface through the AudioPolicyClientInterface::setParameters()
method.
SIP calls should set the audio mode to MODE_IN_COMMUNICATION,
Audio mode MODE_IN_CALL is reserved for telephony.
SIP: Enable built-in echo canceler if available.
1. Always initialize AudioRecord with VOICE_COMMUNICATION.
2. If echo canceler is available, disable our echo suppressor.
Note that this CL is intentionally not correcting the
getAudioSourceMax() return value in MediaRecorder.java as the
new source is hidden here.
Change-Id: Ie68cd03c50553101aa2ad838fe9459b2cf151bc8
udioFlinger.cpp
udioHardwareInterface.cpp
udioPolicyManagerBase.cpp
|
| 61dca6702ce50d937fc764c55bb0842b40dcd0e4 |
25-Jan-2011 |
Jean-Michel Trivi <jmtrivi@google.com> |
Bug 3376700 Add support in APM for stream-specific volume curves
The stream volume was handled the same way for all different stream,
the only potential difference between each of them being the number
of steps available to the user to change the volume. This was
mapped to 99 steps of 0.5dB amplitude, offering a maximum attenuation
of -49.5dB.
This change consists in defining for each stream a curve with two
knees (3 segments) for conversion from volume index to attenuation.
This curve is defined in the AudioPolicyManager in
initializeVolumeCurves(), and can therefore be overridden by the
platform.
Note that this change doesn't modify the volume curves: this CL
enables the curves to be changed by overriding this default
behavior.
Change-Id: I575b66799c52df2906db248943b15120b8a79ea2
udioPolicyManagerBase.cpp
|
| b517e006b51c95db9c44bf5f104f562d431ae597 |
20-Jan-2011 |
Eric Laurent <elaurent@google.com> |
do not merge - Fix issue 3371096.
EffectModule::process() was copying effect chain input buffer to
output buffer if no effect was active instead of accumulating it.
Change-Id: If4ca75601ea69a088d0f71d88aec53e90a1dec89
udioFlinger.cpp
|
| 73337489229cc9b50371c7a9fcd86e9f00ea46d0 |
20-Jan-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 3371096.
EffectModule::process() was copying effect chain input buffer to
output buffer if no effect was active instead of accumulating it.
Change-Id: I2838af2e7b6654d0a76547625929a5453da68d02
udioFlinger.cpp
|
| 935752053ef2691dbb6d5a6d149e0e362c6e3c74 |
19-Jan-2011 |
Eric Laurent <elaurent@google.com> |
Tentative fix for issue 3362362.
The problem is likely that one method is called on the AudioPolicyManagerBase
instance while it is still being constructed by AudioPolicyService.
To avoid this, the AudioPolicyService mutex is held by the constructor until the
platform specific AudioPolicyManager is constructed and the member
mpPolicyManager initialized.
Also added an initCheck() method to AudioPolicyInterface to verify successful
initialization of AudioPolicyManager.
A similar change is done in AudioFlinger constructor.
Also added some missing protections in AudioFlinger methods where the
playback thread list is parsed.
Change-Id: I006b244ec057e1bb0aa5ebe426ef006e3b171056
udioFlinger.cpp
udioPolicyManagerBase.cpp
udioPolicyService.cpp
|
| eb0d0c48ebfcee5e8141ae25fe42a9eac6aae230 |
10-Jan-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 3317627.
The fix consists in selecting the digital audio device (SPDIF/HDMI)
when available if the routing strategy is STRATEGY_PHONE.
Change-Id: Ie500ae92f5c01f2511988543852ba559c6e5994b
udioPolicyManagerBase.cpp
|
| 626c85f3a9ec8e34a1e4af31eddf118d75ee6416 |
17-Dec-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 3217707.
The problem is that when the A2DP headset is disconnected, there is a transition
period during which the A2DP sink pumps data at a very high pace.
This makes that:
1 the audio flinger mixer thread spins and starves binder threads thus delaying
the completion of the A2DP output stream shutdown
2 we read the audio http audio stream faster than normal and we reach the end of stream
for audio while video is still playing if the streamed file is small enough.
The fix consists in detecting abnormal short write intervals and sleep to restore
a normal write pace.
Change-Id: Iab127882494ab0e26266371dc0ce5c2ff6fa476e
2dpAudioInterface.cpp
2dpAudioInterface.h
|
| 0b456a6a33de8cb7c641724579382a31a1beaa2c |
16-Dec-2010 |
Eric Laurent <elaurent@google.com> |
Fix speakerphone routing to analog dock
The audio routing policy when speakerphone is on and a dock with built-in
speakers is connected should be to output audio to teh dock speakers
Also removed route to SCO car kit if forced usage is not SCO as the SCO
socket might not be established.
Change-Id: I1aa2954092e28de935304b90f7a7a64d661934c7
udioPolicyManagerBase.cpp
|
| f3a8d32fd475ec68d966fa6f12a4c975ba80b659 |
15-Dec-2010 |
Eric Laurent <elaurent@google.com> |
Change audio routing policy for HDMI
HDMI device should have a higher priority than analog dock audio but a lower priority
than wired headsets.
Also modified AudioService so that HDMI is mapped to DEVICE_OUT_AUX_DIGITAL device and not
DEVICE_OUT_DGTL_DOCK_HEADSET as before to enable discrimination between SPDIF going to
digital dock and SPIDF going to HDMI.
Change-Id: I887d0c73479784dd2edaf41ce1a7d8d0bdcbb4bd
udioPolicyManagerBase.cpp
|
| b3fd981d5f8bbf629dd8a07b06b4b656b70cb944 |
10-Dec-2010 |
Eric Laurent <elaurent@google.com> |
Fix audio mode log.
Since the new audio mode IN_COMMUNICATION was added, the audio mode log
was broken.
Change-Id: I4fdafc3b98a1b0ceb55058a9e47fed99b3dbe6ad
udioHardwareInterface.cpp
|
| ed7c671a57f89d389c9d6cd67e0698f160f69916 |
01-Dec-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 2641884: Bluetooth volume is dependent on in call volume.
The problem is that the code in AudioPolicyManagerBase::checkAndSetVolume() that forces
voice volume to max when setting bluetooth SCO volume is not called if the bluetooth stream
volume did not actually change. So even if we re apply volumes when switching to bluetooth
device, the volume voice volume is not changed and remains what it was when routed to earpiece
What makes things worse on Passion is that stream volumes are limited when connected to bluetooth
and their actual value does not change as soon as they exceed the limit threshold.
Change-Id: Id7c317db45b392a1c20dca2859678e3c64a371ed
udioPolicyManagerBase.cpp
|
| 094b148b8587a67c19585e2af2cc7b53868bfb89 |
01-Dec-2010 |
Eric Laurent <elaurent@google.com> |
resolved conflicts for merge of 0d28be68 to master
Change-Id: Iec5f810c366d3e1c14a6f6294b0aea4ffb30ae3e
|
| 075a1f660b77fe47b165db1aff97cdd05e300391 |
02-Nov-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 3142808.
There is a bug in the way audio policy manager handles A2DP interface suspend/restore
when SCO is used. This bug is not new but has been triggered by a change in the timing
of the events received by audio policy manager when a call is setup and torn down
introduced by commit 164a8f86c7e48992691368c4895709c3bdb835a4.
The fix consists in grouping the control of A2DP suspended state in a single function
that is called systematically when conditions affecting this state are changed:
- call state change
- device connection/disconnection
- change in forced usage.
Change-Id: I46ee2399ee5547b60511fc6cfd32e2720091b0f8
udioPolicyManagerBase.cpp
|
| 32cb1ba657a1094bb815f4d39f3e4bf12ce8437e |
24-Nov-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 3225810.
Take a wake lock whenever A2DP output stream is active.
Change-Id: Ie50e6d4cb34c8a1ba97b301ef25e10aeb153d8f3
2dpAudioInterface.cpp
2dpAudioInterface.h
|
| f5aafb209d01ba2ab6cb55d1a12cfc653e2b4be0 |
18-Nov-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 3157123.
Use a Mutex wherever atomic operations were used in AudioTrack,
AudioRecord, AudioFlinger and AudioEffect classes.
Change-Id: I6f55b2cabdcd93d64ef19446735b8f33720f8dbc
udioFlinger.cpp
udioFlinger.h
|
| f1fb01a7f00b8da90a36268aba8584a872e99175 |
15-Nov-2010 |
Jean-Michel Trivi <jmtrivi@google.com> |
Add new audio mode for audio communications other than telelphony.
The audio mode MODE_IN_CALL signals the system the device a phone
call is currently underway. There was no way for audio video
chat or VoIP applications to signal a call is underway, but not
using the telephony resources. This change introduces a new mode
to address this. Changes in other parts of the system (java
and native) are required to take this new mode into account.
The generic AudioPolicyManager is updated to not use its phone
state variable directly, but to use two new convenience methods,
isInCall() and isStateInCall(int) instead.
Change-Id: Id744cd26520ea1d1a4795eabe6a1f0c58789af76
udioFlinger.cpp
udioPolicyManagerBase.cpp
|
| 56ecd20263d7f63476f756fc5d8b043b325c7bfb |
09-Nov-2010 |
Jean-Michel Trivi <jmtrivi@google.com> |
Add support for audio recording source in generic audio policy mgr.
Update the platform-independent audio policy manager to pass the
nature of the audio recording source to the audio policy client
interface through the AudioPolicyClientInterface::setParameters()
method.
Change-Id: I6b4fd0f8a3acea0d7d30bbad98edd1977dc012bf
udioPolicyManagerBase.cpp
|
| c643d77a099b0e86ad12de9b498f6f780e03184e |
09-Nov-2010 |
Jean-Michel Trivi <jmtrivi@google.com> |
Add recording source for voice communication
Add a recording source used to designate a recording stream for
voice communications such as VoIP.
Change-Id: I4091d67069b1a0170c1a5ca5e6acd51eb0aa08f9
udioPolicyManagerBase.cpp
|
| b235dee7be544a6c990ba4f8d89ab03e38c826f8 |
06-Oct-2010 |
Praveen Bharathi <pbharathi@motorola.com> |
Added support for dock headset observer
Change-Id: I06b2e65e3bfa10735e6c7fd3349afa9ae7d45292
Signed-off-by: Praveen Bharathi <pbharathi@motorola.com>
udioPolicyManagerBase.cpp
|
| 173e75df12ad5398f3650a30e430dcac0cbcf278 |
07-Oct-2010 |
Eric Laurent <elaurent@google.com> |
am ce2e2184: am 37947afe: Merge "Fixed AudioFlinger not always pausing tracks" into gingerbread
Merge commit 'ce2e2184bbc5530f4fac3220fdf3d1b3fc08a4c3'
* commit 'ce2e2184bbc5530f4fac3220fdf3d1b3fc08a4c3':
Fixed AudioFlinger not always pausing tracks
|
| af59ce2407fa4e6e5d8f1664a4df2daf1badd407 |
05-Oct-2010 |
Eric Laurent <elaurent@google.com> |
Fixed AudioFlinger not always pausing tracks
If the pause request is received before the AudioTrack buffer was
completelly filled and the track ready for mixing, the pause is
not executed: the track just underruns and stays in pausing state.
The fix consists in considering the track ready for mixing immediately
if pausing.
Change-Id: Ia6cb4703fee2126e41011a6400ea8eeb3a3e5456
udioFlinger.cpp
|
| 9ad6e8bd9fab860e639f6ce95479e4b2a9b03ff5 |
01-Oct-2010 |
Eric Laurent <elaurent@google.com> |
am de12c3cf: am 220ab887: Merge "Issue 3032913: improve AudioTrack recovery time" into gingerbread
Merge commit 'de12c3cf56e3f27b2efc60eeae8b5e422747f2b9'
* commit 'de12c3cf56e3f27b2efc60eeae8b5e422747f2b9':
Issue 3032913: improve AudioTrack recovery time
|
| 44d9848d6656777a18019223e0d35f2fcc67719a |
01-Oct-2010 |
Eric Laurent <elaurent@google.com> |
Issue 3032913: improve AudioTrack recovery time
This issue showed that when an AudioTrack underruns during a too long period
of time and is therefore disabled by audioflinger mixer, it takes an additional
delay of up to 3 seconds to recover.
This fix adds a simple mechanism to recover immediately when the client application
is ready to write data again in the AudioTrack buffer
Also throttle warnings on record overflows
Change-Id: I8b2c71578dd134b9e60a15ee4d91b70f3799cb3d
udioFlinger.cpp
|
| 9e2e2d2bc86372bde4673c31ef416b2f29d87ac2 |
29-Sep-2010 |
Eric Laurent <elaurent@google.com> |
am b047e3cd: am aeb2c62e: Merge "Fix several audio effects problems." into gingerbread
Merge commit 'b047e3cdf24b32e26f366fcd4cd0eee8ae6c592c'
* commit 'b047e3cdf24b32e26f366fcd4cd0eee8ae6c592c':
Fix several audio effects problems.
|
| dac69110ed1073bf0a9827a3f78698896dd05d97 |
28-Sep-2010 |
Eric Laurent <elaurent@google.com> |
Fix several audio effects problems.
Fixed the following issues in LVM effect bundle wrapper:
- memory leaks in EffectCreate() in case effect creation fails at various stages
- Added saturation when accumulating to output buffer
- Fixed problems with enabled effects count when an effect is released while enabled
- Do not allocate temporary buffer for accumulation each time process() is called
Fixed the following issues in effects framework (AudioFlinger)
- Release effect synchronously in the library when deleted from effect chain
- Do not call the effect process function if no tracks are present in the same
audio session
Change-Id: Ifbd80a163415cfb3c0a337c12082853ea45d9c91
udioFlinger.cpp
|
| f43c92f8d7b1d81cc7248031970d68dc65579be0 |
24-Sep-2010 |
Eric Laurent <elaurent@google.com> |
am 692dfafe: am 880dfe4f: Merge "Fix issue 3007862" into gingerbread
Merge commit '692dfafe02d04cdbab5367546e166580c92e4d2e'
* commit '692dfafe02d04cdbab5367546e166580c92e4d2e':
Fix issue 3007862
|
| 84e9a10fde8a4ae3da4f88d7911c154933aa457f |
24-Sep-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 3007862
Removed a cross deadlock condition between audioflinger and audio policy
service mutexes.
Audioflinger::createEffect() locks audioflinger mutex and then calls
AudioSystem::getOutputForEffect() which ends up in
AudioPolicyService::getOutputForEffect() which locks audio policy service
mutex. If at the same time, the command thread in audio policy service is
processing a command(set volume, set route...), the mutex is locked and the
command will call one audioflinger method which in turn will attempt to
lock audioflinger mutex.
The fix consists in releasing audioflinger mutex before calling
getOutputForEffect().
Change-Id: Id44e7feb36e0a295731f6aa97cf32d022edd34d0
udioFlinger.cpp
|
| cc08406f74b67b1b7e532ef8ed534c0476ced636 |
23-Sep-2010 |
Eric Laurent <elaurent@google.com> |
am 11746caa: am 08959c63: Merge "Request permission for global audio effects." into gingerbread
Merge commit '11746caaa852984ff186bf5b8807e2c14cd7c1bc'
* commit '11746caaa852984ff186bf5b8807e2c14cd7c1bc':
Request permission for global audio effects.
|
| fac4895de4ae63928ff0cf2ccece106eb6d33f72 |
22-Sep-2010 |
Eric Laurent <elaurent@google.com> |
Request permission for global audio effects.
Applications creating an audio effect on the output mix must
have the MODIFY_AUDIO_SETTINGS permission.
Change-Id: I57d88533f91ad0d33680107d79abcec28f7263b5
udioFlinger.cpp
|
| d20b4cfde2848052cf93ed13df14d75347ec64fd |
13-Sep-2010 |
Eric Laurent <elaurent@google.com> |
am 4d987850: am bd2e9ec6: Merge "Fix volume problems with insert revert" into gingerbread
Merge commit '4d9878502f7661ed34540a485a5942d859e209c7'
* commit '4d9878502f7661ed34540a485a5942d859e209c7':
Fix volume problems with insert revert
|
| e0aed6ddcb4e3c301b80aa26706b6052dab42c41 |
11-Sep-2010 |
Eric Laurent <elaurent@google.com> |
Fix volume problems with insert revert
- Use a constant input level to the reverb engine and implement volume control in the
insert reverb. This avoids the volume spikes when an effect that was inserted after
the reverb is disabled or removed.
- Fix clicks (one silent buffer) at the end of the reverb disable period.
- Modified volume management in audioflinger so that the volume ramp is also done by
the insert effect if present when the track is paused (avoids clicks).
- Increased room level for all presets.
Also fixed problems with output stage session (-1):
- effect bundle wrapper was not designed to support session -1
- the permission check in audioflinger for using session -1 failed due to a wrong usage of
getCallingPid()
Change-Id: Id1ff51327263364bf71d3f2668fa5cde4311d84f
udioFlinger.cpp
|
| e5fe2601b25e3220c3c4cfb79695ddb8a2f817ac |
02-Sep-2010 |
Eric Laurent <elaurent@google.com> |
am 937056c1: am 32ec1ad1: Merge "Fix problem in AudioEffect::command() status." into gingerbread
Merge commit '937056c1a7ce71e6881305a31f76e62a76685f24'
* commit '937056c1a7ce71e6881305a31f76e62a76685f24':
Fix problem in AudioEffect::command() status.
|
| aeae3de947fa0b1e670c8472b32288962f97b4f5 |
02-Sep-2010 |
Eric Laurent <elaurent@google.com> |
Fix problem in AudioEffect::command() status.
The *pReplyData argument of the command() function was left unitialized by EffectHandle::command()
when command was EFFECT_CMD_ENABLE, EFFECT_CMD_DISABLE and EFFECT_CMD_SET_PARAM_COMMIT.
Change-Id: I91a19817ead2a8cfbdd8e2d77ca270c7ce9d5bd4
udioFlinger.cpp
|
| 1040127473ee48c1c3b8c279ae05d38dbfb6e4c2 |
01-Sep-2010 |
Eric Laurent <elaurent@google.com> |
am 55e79374: am f560ceab: Merge "Audio Effects: fix problems in volume control." into gingerbread
Merge commit '55e7937462d107b469a58651d3d51e15086f8e9e'
* commit '55e7937462d107b469a58651d3d51e15086f8e9e':
Audio Effects: fix problems in volume control.
|
| 8f45bd725549436eeacd12ee69349e2332ed8da5 |
31-Aug-2010 |
Eric Laurent <elaurent@google.com> |
Audio Effects: fix problems in volume control.
- Fixed click when re-enabling effect during the turn off phase:
make sure the effect states where effect is processed are the same
where volume control is delegated to effect.
- Fixed click when effect is deleted while still active: do not apply
volume ramp if an effect having volume control was just removed from the
effect chain.
Also fixed a crash when PCM dump is enabled in effect bundle wrapper.
Change-Id: Ib562f5cf75c69af75df0e862536262e2514493e4
udioFlinger.cpp
udioFlinger.h
|
| ea798dd9f37b34acfa3cf8d945dbe7f7e90e7196 |
31-Aug-2010 |
Eric Laurent <elaurent@google.com> |
am 44217848: am ae6bdc23: Merge "Fix issue 2952766." into gingerbread
Merge commit '4421784895a58bb7bcf90236a9e443b372b5b80e'
* commit '4421784895a58bb7bcf90236a9e443b372b5b80e':
Fix issue 2952766.
|
| c1c88e27a6f07a530fca184f515c16d51e46d56f |
28-Aug-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 2952766.
The problem is that the audio policy manager does not handle the input devices
when forced use for telephony is changed.
The problem does not appear in a call over PSTN becasue only teh output devices drives the
routing of in call audio to/from the base band.
The fix consists in modifying AudioPolicyManagerBase::setForceUse() to check for active inputs
and update the input device if needed.
Change-Id: I0d36d1f5eef1cce527929180c29b025439902f10
udioPolicyManagerBase.cpp
|
| 69e0b8c5556df1d4db1a730db1c13409a96749c7 |
21-Aug-2010 |
Eric Laurent <elaurent@google.com> |
am 3a899b85: am 0b9a12bf: Merge "Fix issue 2929440" into gingerbread
Merge commit '3a899b85102f6d51585b3d46ced3038c8602f202'
* commit '3a899b85102f6d51585b3d46ced3038c8602f202':
Fix issue 2929440
|
| 25cbe0ecd6df8be7e40537c5d85c82f105038479 |
19-Aug-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 2929440
Fixed regression introduced by change a54d7d3d7dd691334189aab20d23c65710092869 in audioflinger mixer thread:
When the output stream is suspended, the sleep time between two writes must match the actual duration
of one output stream buffer otherwise the playback rate is not respected.
Change-Id: Ic5bebe890290d1f44aeff9dd3c142d18e26fff2a
udioFlinger.cpp
udioFlinger.h
|
| a1a6cba3664a04d214b372526658ba227e4d5366 |
14-Aug-2010 |
Eric Laurent <elaurent@google.com> |
am f28608cd: am c8d0d630: Merge "Fix issue 2909189: System property ro.audio.silent no longer mutes system." into gingerbread
Merge commit 'f28608cdf4f6a0075d0811ccd32123baf4daec31'
* commit 'f28608cdf4f6a0075d0811ccd32123baf4daec31':
Fix issue 2909189: System property ro.audio.silent no longer mutes system.
|
| 571d49c1c316f5e07b74ed7b5df6bdec7cbc1a14 |
11-Aug-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 2909189: System property ro.audio.silent no longer mutes system.
Fixed regression introduced by commit 2a6b80bc65c4782b5a7168b300e1dc5ec9f617ee:
master mute was not working if no effect chains were present on session 0.
Change-Id: I66d107e045d159cb94d29c7476fa1e12d92f2ae7
udioFlinger.cpp
|
| bc417c0531332635df9525df4683274a08fb987f |
10-Aug-2010 |
Eric Laurent <elaurent@google.com> |
am 1ece7d68: am 2a6b80bc: Fixed several audio effects problems.
Merge commit '1ece7d6873027068a3e7b055597045e9387b217c'
* commit '1ece7d6873027068a3e7b055597045e9387b217c':
Fixed several audio effects problems.
|
| 8569f0d3bf4c6787707e348a7cf73b9c4199cb32 |
30-Jul-2010 |
Eric Laurent <elaurent@google.com> |
Fixed several audio effects problems.
- Fixed constant inversions in AudioEffect.java
- Do not return error when enabling an already enabled effect
- Update cached effect state in native AudioEffect class when effect is enabled/disabled by command() method
- Remove click when restarting effect during disable sequence
- Fixed problem in master mute management when volume control is delegated to effect.
Change-Id: I6df4ce9fcc54fdc7345df858f639d20d802d6712
udioFlinger.cpp
|
| e679114b6c7e087a59c2b2c3b55603c43da4ef38 |
29-Jul-2010 |
Eric Laurent <elaurent@google.com> |
am ba676a00: am a54d7d3d: Fixed underrun in audioflinger mixer.
Merge commit 'ba676a00c1b9ba59b30fe805b83d03d1768fffbd'
* commit 'ba676a00c1b9ba59b30fe805b83d03d1768fffbd':
Fixed underrun in audioflinger mixer.
|
| 60e182437228312cc28469a5b0dfde77ac848e1a |
29-Jul-2010 |
Eric Laurent <elaurent@google.com> |
Fixed underrun in audioflinger mixer.
When all audio tracks have been disabled and the mixer is running idle before the output stream is placed in standby,
the mixer sometimes fails to write to the output stream on time to avoid underrun.
This is because the sleep period used to wait before the next write to output stream is too close to the actual buffer duration.
In fact this sleep time is not critical as if we write too early to the output stream, the kernel driver will wait for free buffers
from the audio DSP DMA and we will sleep anyways.
The fix consists in dividing the calculated wait period by 2 to increase the margin.
Change-Id: I5730887dc2ccce2a511bc858494a6f7da6b392a0
udioFlinger.cpp
|
| cc759794489c2c39aaef229fba4ace32e394bb81 |
29-Jul-2010 |
Eric Laurent <elaurent@google.com> |
am 45dc4f82: am 493941b8: Allow creation of an audio effect on a session with no audio tracks.
Merge commit '45dc4f82a00e52b12389b22a7cfbbee5609e8e28'
* commit '45dc4f82a00e52b12389b22a7cfbbee5609e8e28':
Allow creation of an audio effect on a session with no audio tracks.
|
| 39e94f8f723d445447fdee0822291e664b631f60 |
28-Jul-2010 |
Eric Laurent <elaurent@google.com> |
Allow creation of an audio effect on a session with no audio tracks.
This is necessary to allow creating and enabling an effect attached to a particular player
session before the playback is started. As a matter of fact, the implementation of the mediaplayer
does not create the AudioTrack before playback starts.
Change-Id: I1266e8885f9d756acc949303321aaac0fbf83e34
udioFlinger.cpp
udioFlinger.h
|
| ed0de3c123aaa86147993721d4794648a65aab75 |
28-Jul-2010 |
Eric Laurent <elaurent@google.com> |
am 6fa88882: am 62d83a0c: Merge "Audio effects: modified command() parameter types." into gingerbread
Merge commit '6fa8888254deb4961dc64c6ac03dc103c9cd8bcf'
* commit '6fa8888254deb4961dc64c6ac03dc103c9cd8bcf':
Audio effects: modified command() parameter types.
|
| 25f4395b932fa9859a6e91ba77c5d20d009da64a |
28-Jul-2010 |
Eric Laurent <elaurent@google.com> |
Audio effects: modified command() parameter types.
The type of the cmd, cmdSize and *pReplySize parameters of the effect control interface command()
function have been modified from int to uint32_t. This is more consistent with their role.
Change-Id: I84d289fc262d6753747910f06f485597dfee6591
udioFlinger.cpp
udioFlinger.h
|
| 8184a5734690b30f4de0f6b6c16daf1e089f67df |
20-Jul-2010 |
Eric Laurent <elaurent@google.com> |
resolved conflicts for merge of dd206093 to master
Change-Id: I21dd2321a4839d034d49092baccbf40986f17dae
|
| de070137f11d346fba77605bd76a44c040a618fc |
13-Jul-2010 |
Eric Laurent <elaurent@google.com> |
Audio policy manager changes for audio effects
Added methods for audio effects management by audio policy manager.
- control of total CPU load and memory used by effect engines
- selection of output stream for global effects
- added audio session id in parameter list for startOutput() and stopOutput().
this is not used in default audio policy manager implementation.
Modifications of audio effect framework in AudioFlinger to allow moving and reconfiguring
effect engines from one output mixer thread to another when audio tracks in the same session
are moved or when requested by audio policy manager.
Also fixed mutex deadlock problem with effect chains locks.
Change-Id: Ida43484b06e9b890d6b9e53c13958d042720ebdb
udioFlinger.cpp
udioFlinger.h
udioPolicyManagerBase.cpp
udioPolicyService.cpp
udioPolicyService.h
|
| bbf6dcd2aa8eb877e6412aaade25a077f62a6284 |
19-Jul-2010 |
Eric Laurent <elaurent@google.com> |
am 3c6ea955: am 7203272f: Merge "Fixed problems in audio effect volume control." into gingerbread
Merge commit '3c6ea95553d3a0e416874c583a41efff3f6fbc57'
* commit '3c6ea95553d3a0e416874c583a41efff3f6fbc57':
Fixed problems in audio effect volume control.
|
| f997cabca292d70d078ae828e21c28e6df62995f |
19-Jul-2010 |
Eric Laurent <elaurent@google.com> |
Fixed problems in audio effect volume control.
Fixed the following problems in audio effect volume control in AudioFlinger:
- Make sure that the volumes returned by EffectChain::setVolume_l() are correct even is
no change is detected since last call
- Do not use isEnabled() to validate volume control but mState >= ACTIVE instead as the volume control
must be also active in STOPPING and STOPPED states.
Change-Id: Id62da3164fad500ee8a5efd6cd78c77e8fdcb541
udioFlinger.cpp
udioFlinger.h
|
| 521bcb0ff5054e5be40f5194026da3463504ae3b |
16-Jul-2010 |
Eric Laurent <elaurent@google.com> |
am f16b3f10: am b0374a0e: Merge "Several improvements in audio effects volume control." into gingerbread
Merge commit 'f16b3f104f60738447db60bcf2c44c7610e17199'
* commit 'f16b3f104f60738447db60bcf2c44c7610e17199':
Several improvements in audio effects volume control.
|
| cab112421da6e8eac19ffddbbe3d76067cffee78 |
15-Jul-2010 |
Eric Laurent <elaurent@google.com> |
Several improvements in audio effects volume control.
- Fixed crash when deleting an effect chained before an effect having volume control
- Changed EFFECT_FLAG_VOLUME_CTRL to implicitely include EFFECT_FLAG_VOLUME_IND
(not need to set both in effect descriptor).
- Volume control changes from one effect to another if needed according to effect enable state
- EFFECT_CMD_SET_VOLUME is only sent when their is an actual change in volume
Change-Id: Ieebaf09157e2627366023569d95516646e03e26c
udioFlinger.cpp
udioFlinger.h
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| c0b9c87b2581af430e143822918935d15ac69911 |
15-Jul-2010 |
Mathias Agopian <mathias@google.com> |
am c74034b6: am 24651682: added BinderService<> template to help creating native binder services
Merge commit 'c74034b6bf0940dc80c4d4efa1e677ad0d2dfd6c'
* commit 'c74034b6bf0940dc80c4d4efa1e677ad0d2dfd6c':
added BinderService<> template to help creating native binder services
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| 094c96da874d84eae46f76cc036fa7b7993eb999 |
15-Jul-2010 |
Mathias Agopian <mathias@google.com> |
am 78983a91: am 4055b780: Merge "move native services under services/" into gingerbread
Merge commit '78983a9133d3dd3f08b1ec462a7e2f9e7bfa9e2f'
* commit '78983a9133d3dd3f08b1ec462a7e2f9e7bfa9e2f':
move native services under services/
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| 5462fc9a38fa8c9dff434cd53fa5fb1782ae3042 |
15-Jul-2010 |
Mathias Agopian <mathias@google.com> |
added BinderService<> template to help creating native binder services
Change-Id: Id980899d2647b56479f8a27c89eaa949f9209dfe
udioFlinger.cpp
udioFlinger.h
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| 65ab47156e1c7dfcd8cc4266253a5ff30219e7f0 |
15-Jul-2010 |
Mathias Agopian <mathias@google.com> |
move native services under services/
moved surfaceflinger, audioflinger, cameraservice
all native services should now reside in this location.
Change-Id: Iee42b83dd2a94c3bf5107ab0895fe2dfcd5337a8
2dpAudioInterface.cpp
2dpAudioInterface.h
ndroid.mk
udioBufferProvider.h
udioDumpInterface.cpp
udioDumpInterface.h
udioFlinger.cpp
udioFlinger.h
udioHardwareGeneric.cpp
udioHardwareGeneric.h
udioHardwareInterface.cpp
udioHardwareStub.cpp
udioHardwareStub.h
udioMixer.cpp
udioMixer.h
udioPolicyManagerBase.cpp
udioPolicyService.cpp
udioPolicyService.h
udioResampler.cpp
udioResampler.h
udioResamplerCubic.cpp
udioResamplerCubic.h
udioResamplerSinc.cpp
udioResamplerSinc.h
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